Interconnect

Thursday, April 05, 2012

Science Or Snake Oil? The Facts Behind The Hype About Loudspeaker Wire

Marketers must come up with reasons for why you should buy their wire. To claim that their wire is better, they must first identify, in some cases invent, a difference

Too many good folks have been separated from their hard earned money by hyperbolic claims about loudspeaker wire. There will always be people with more dollars than sense, but they don’t last very long in professional audio.

I speculate there aren’t many (if any) of you who would pay thousands, or even tens of dollars per foot for speaker wire.

A very basic practice in merchandising is called differentiation. Marketers must come up with reasons for why you should buy their wire. To claim that their wire is better, they must first identify, in some cases invent, a difference.

This search for a selling proposition has sometimes focused on “skin effect.” It’s a real effect and describes how at very high frequencies, electrons travel in the outer layer or “skin” of signal conductors.

Another related property is that high frequency signals travel faster than low frequencies through the same cable.

These phenomena are dealt with appropriately in very high frequency applications with several techniques. For example, one particular type of botique wire is made up of a large number of very small conductors braided or woven into one cable, producing a large surface area or “skin” for a given cross sectional area.

Another approach for high power high frequency power transfer is to use a hollow conductor, resembling a section of copper tubing. If the electrons are going to ignore the center of the conductor, why pay for it?

This is not an issue for audio professionals, working at mere audio frequencies of 20 Hz to 20 kHz. Perhaps it would be if we were sending audio over many miles, like the telephone company in its pre-digital days. They had to periodically correct for waveform smear.

But at the speed that electricity travels, our typical path distances are much too short to be an issue.

Out Of Perspective
Wire is not very sexy or easy to create real marketing hooks for, but it can actually make an audible difference. The dominant mechanism is simple resistance.

It’s perhaps ironic that the “snake oil” markers of loudspeaker wire will exaggerate some real but insignificant parameter far out of perspective while compromising the real deal.

Forget the hype, what’s important for loudspeaker wire is that it exhibit low impedance that is resistive in nature. If the wire has a significant impedance component (reactance) that changes over the audio frequency spectrum, this can form a simple divider with the loudspeaker’s resistive impedance and cause a frequency response error.

In addition, since loudspeaker impedance will vary quite a bit over frequency, even a perfectly resistive speaker wire will cause errors. The magnitude of this frequency response error will increase proportionately as the wire’s resistance increases.

Purveyors of “funny wire” don’t bother to make claims about useful metrics like resistance since that is already defined by the wire size or gauge (known as “American Wire Gauge” or AWG for short). That would be like advertising how many quarts were in their gallons!

However, frequency response errors caused by wire resistance are one of the very real things that people actually do hear. I find this following anecdote instructive.

From a discussion with one individual who was certain that he heard a significant improvement when using his “Snake-O Special” loudspeaker wire (name changed because I don’t remember it), I determined that the wire gauge he was using was marginal for the length of his run. The wideband loss of volume caused by a wire’s resistance will be very difficult to hear without a side-by-side comparison.

But the difference in amount of loss caused by the loudspeaker’s changing impedance at different frequencies can easily cause a frequency response error that is probably what he heard. It’s easy to imagine how a rising impedance at high frequency could cause a pleasant sounding treble boost. Just listen to how clean and clear these “Snake-O Specials” sound!

There are several strategies to manage these real losses from wire resistance. The obvious one is to throw more copper at the problem. Heavier gauge wire with lower resistance will exhibit lower losses for a given run length.

Another fairly obvious approach is to locate the amplifiers as close as possible to the loudspeakers to keep the run length as short as possible. A third less obvious approach is to scale up the intermediate signal voltages.

Constant Voltage
There are cases, such as in large distributed sound systems where neither of the first two approaches is cost effective.

You can’t afford to put a separate amplifier at every loudspeaker location, and sending sound sources over long distances with acceptable losses would require very heavy gauge wire.

The solution borrows a strategy from high voltage power distribution systems such as the one used by utilities to bring electrical power to our homes.

The power developed within a given load increases with the square of the terminal voltage (E^2/R). However, wire’s losses only increase linearly with current flow, because the voltage developed across the wire is a simple function of its resistance times that current. 

Power engineers determined that by raising the voltage carried by transmission lines they could increase the power being carried exponentially while simultaneously reducing the losses due to current flow.

The utility company accomplishes this magic with step-up/step-down transformers. By “transforming” a typical 100-amp at 240-volts residential service, up to tens of thousands of volts at the transmission line the 100-amp draw is reduced to the far more manageable level of 1 amp or so. Wire losses are 1 percent of what they would otherwise be.

Similar manipulations go on in “constant voltage” distributed sound systems but rather than stepping up the voltage to thousands of volts the standard for U.S. systems is 70-volt, with Europe using a slightly higher 100-volt standard. The rest of the world tries to conform to one of those two standards.

Of course, the audio signal isn’t actually held constant. The voltage at rated power is. Both 5 watts and 500 watts constant voltage systems deliver the same nominal voltage for distribution.

The goal in any effective distribution system is to deliver as much power as possible to do useful work in the load and waste as little as possible heating up the wire. In a simple distributed sound system sending a few watts of announcements across a few hundred feet of factory floor, the typical low voltage system could drop as much power in the speaker wire as would reach the loudspeakers.

By stepping up to 70 volts and back down again at each loudspeaker the balance of power delivered versus lost is more respectable.

To put numbers to this concept, say we are trying to deliver 1 watt each to two loudspeakers located 100 feet distant from an amplifier using 24 AWG wire. Because we must count wire losses from the feed coming and going, 200 feet total of 24 AWG exhibits resistance of approximately 5 ohms.

Figure 1: Two different ways of realizing one watt at two loudspeakers. Click to enlarge.

To realize 1 watt at each loudspeaker, there would need to be more than 4 watts into the wire at the amplifier end. (Over 2 watts gets wasted as heat in the wire).

 
If we first step up the audio to a nominal 70-volt level the current drops to such a low level that the same wire would only waste 0.14 watts while delivering the same 1 watt each to the two loudspeakers.

As useful as constant (high) voltage systems are for managing wire losses, they don’t make sense for point-to-point runs in sound reinforcement systems. The main drawback is the size of the step-up and step-down transformers required.

To put this in perspective, the size of the transformer has to double every time you drop the frequency an octave. To cleanly pass 20 Hz both step-up and step-down audio transformers would have to be three times the size of a conventional amplifier’s 60 Hz power supply transformer.

Keep It Short
The good news for most live sound applications is that we don’t have to tolerate extremely long wire runs. By locating power amplifiers near the loudspeakers we can keep wire runs reasonably short. At these shorter distances we can easily afford heavier gauge wire.

While power losses are now manageable, it is worthwhile investigating the next dominant consideration in sizing loudspeaker wire.

Frequency response errors will be caused by the voltage divider created between the wire’s fixed resistance and the loudspeakers changing impedance versus frequency.

Figure 2 and Figure 3 show two representative loudspeaker impedance plots, pulled from the Internet.

These are not offered as either worst case or typical.

From the impedance plot in Figure 2, if we ignore the extreme low frequency, this loudspeaker exhibits a maximum impedance greater than 17 ohms, with a significant region of the upper bass down around five ohms.

Figure 2: This loudspeaker exhibits a maximum impedance greater than 17 ohms. Click to enlarge.

Meanwhile, Figure 3, while more complex, covers a similar impedance range, with a maximum around 16 ohms and a minimum around six ohms.

 
To derive a frequency response error we need to compare the drop at maximum impedance to the drop at minimum impedance. The equations below calculate that drop for a given wire resistance.

Note: To simplify this analysis we will assume all loudspeaker impedances to be resistive. While not strictly accurate, loudspeaker impedances will typically be resistive at impedance minimums and any errors caused by load phase angle at the impedance maximums will not be significant for the sake of this analysis.

Minimum Voltage drop= V max = Z max /(Z max +Z wire) 
Maximum Voltage drop= V min = Z min /(Z min + Z wire)

Frequency Response deviation= FR max = -20 Log10 (V min/ V max)

Solving for 1-, 0.5-, and 0.1-ohm wire resistance we get:

Loudspeaker….......1 ohm…...... 0.5 ohm….. 0.1 ohm

Spkr 1 (17/5)........ -1.09 dB…... -.57 dB…... -.12 dB

Spkr 2 (16/6)....... -.81 dB…..... -.42 dB…... -.09 dB

Figure 3: While more complex than the loudspeaker in Figure 1, this covers a similar impedance range, with a maximum around 16 ohms. Click to enlarge.

Another related consequence is how wire resistance degrades effective damping factor.

While damping factor is usually though of as a power amplifier characteristic, in reality the wire selection can easily dominate actual damping available at the loudspeaker.

In the above examples the 1-ohm wire would by itself cause a rather weak damping factor of 5 or 6 (regardless of the amplifier’s rated damping factor).

Using the 0.1-ohm wire predicts a more respectable 50-60 damping factor, with some small additional degradation due to the amplifier’s output impedance.

Damping factor deserves a more extensive discussion, but for this exercise we will assume that the amplifier’s output impedance is small with respect to our wire’s resistance.

Gauging Gauge
It’s difficult to predict a precise threshold for audibility of frequency response errors.

Controlled listening tests have suggested that differences as small as a few tenths of a dB can be audible.

To satisfy the dual goals of minimizing frequency response errors and not degrading damping factor for the example loudspeakers selected, I am comfortable with targeting a total wire resistance on the order of 0.1 ohm.

Wire’s resistance varies linearly with length. To keep the total resistance below our target limit of 0.1 ohm we must first project the length of our desired wire run, and then select a wire gauge whose resistance per unit length keeps us within the total resistance budget.

Don’t overlook that the wire length is actually twice the run distance as we must consider the feed to and return from the loudspeaker as effectively in series. We must also add in contact resistance for the connections at all ends.

Lets look at how this works out for a practical example of a 20-foot run. First, we double that to 40 feet to establish the true signal path length.

Then we need to account for contact resistance. I’ve seen Neutrik Speakon (or copies of that connector) rated as low as 1mOhm (1/1000th ohm) per contact when new, and guaranteed

< 2 mOhm over life.

Because there are four connections in our total path lets budget .008 ohms for connections. Subtracting this 0.008 ohms from our 0.1-ohm target leaves us .092 ohms for wire. Dividing this 0.092 ohms by the 40-foot length calculates out to 0.0023 ohms per foot.

Plugging this into the equation for wire gauge:

AWG = 10 ×log 10 R +10 (note R is per 1000 feet)

We get:  AWG = 10x log 10 (2.3) +10 = 13.6 gauge

This is a little cumbersome, but once you have established an appropriate gauge for a nominal run length with your specific system. This gauge can be scaled up or down for other run lengths.

Wire resistance changes linearly with length. It changes non-linearly with gauge. A convenient property of wire gauge is that the wire’s resistance will double for every 3-step increase in gauge (AWG). Conversely the resistance will drop in half for a three-step decrease in gauge.

Based on this same example and rounding off to 14 AWG, we can expect similar performance from a 40-foot run using 11 AWG wire, and a 10-foot run would only need 17 AWG. This numbering convention gets a little unusual below “0” AWG.

One step below (larger than) “0” is “00”, and “000” is two steps larger than “0”. I don’t expect to see speaker wires this large, as they would be very difficult to effectively interface with amplifiers and loudspeakers.

Using this example to size wire for your system will get you in the ball park, but it will be more accurate to use actual impedance specifications for your loudspeakers. Manufacturers of professional loudspeakers routinely publish this information.

Remember, use only the impedance max/min deviation within the audio bandwidth of interest. It doesn’t matter what a tweeter’s DC resistance is or a woofer’s 20 kHz impedance, since you won’t be listening to them there.

You also may want to tighten or relax the acceptable frequency response deviation. Better yet, look at your loudspeaker’s typical frequency response and determine if the response errors caused by your wire losses are additive or corrective.

While I don’t suggest trying to dial in corrective equalization using wire losses, if the error is making your system flatter you can afford to be less aggressive in sizing your wire AWG as long as you keep damping and power losses under control.

John Roberts is a long-time professional audio product and system designer and has been writing outstanding technical articles over the past two decades.

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Posted by admin on 04/05 at 01:46 PM
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Balanced Versus Unbalanced Lines

Two flavors of cable, each with their pros, cons, and best applications.

Unbalanced Lines
Unbalanced signal lines are characterized by the fact that the cable and connectors use only two conductors, a center conductor surrounded by a shield.

Examples of unbalanced wiring are found in tip/sleeve 1/4-in guitar cords or the cables used with many CD players and tape decks which terminate with RCA phono type connectors.

In an unbalanced configuration, the shield surrounds a single center conductor.

The shield stays at a constant ground potential (as it is connected to ground when plugged into equipment) while the signal voltage in the center conductor varies in a positive and negative manner relative to it.

Because the shield completely surrounds the center or “hot” conductor and is connected to ground, it intercepts most of the electrical interference encountered by the cable and passes it away harmlessly to ground.

Very little or no interference will be able to reach the center conductor where it would interact with desired signal.

Because the shield is one of the two conductors required to complete the circuit, it must always be connected at both ends of the cable.

This may set up a condition called a “ground loop” that sometimes produces hum when the grounds of different pieces of electrical equipment are connected to each other.

Unbalanced: A single center conductor is surrounded by a shield.

Note: A shield that consists of wire that is braided instead of just spun around the center in a spiral will provide superior coverage. Spiral shield is less expensive but can spread apart when the cable is flexed, exposing the center conductor to unwanted hum and buzz.

If outside electrical interference does manage to penetrate the shield, it will mix with the desired signal that is present in the center conductor and be amplified right along with it as noise, buzz, etc.

This might not be a huge problem with electric guitars, tape decks and unbalanced microphones when the cable is only a few feet long.

But in environments containing a lot of interference or when an unbalanced signal is sent long distances, such as down a snake, it will become more and more susceptible to unwanted interference.

This problem can be alleviated with the use of balanced lines.

Balanced Lines
Balanced lines are characterized by the fact that there are two center conductors for the signal, surrounded by a shield.

This shield is connected to ground like unbalanced lines but it is not required as one of the signal conductors. Its sole purpose is to provide its line of defense against unwanted interference.

A benefit of this configuration is that the shield only needs to be connected to ground at one end of the cable in order for it to work.

Having this ground disconnected or “lifted” at one end can eliminate the ground loop problem discussed in the previous section on unbalanced lines.

Exception: the ground must be connected at both ends when transmitting phantom power. Phantom power will not work if the ground is lifted at either end.

Balanced: Two center conductors are surrounded by a shield.

The two center conductors of a balanced line act as the sole conduit for the signal and operate in a “push-pull” manner.

That is, as the voltage on one conductor becomes positive, the voltage on the other conductor becomes negative by the same amount and at the same time (and vice-versa).

So at any point in time, both conductors are equal in voltage but opposite in polarity.

The receiving circuit that processes this balanced signal is called a differential amplifier and this opposing polarity of the voltages on the conductors is essential for its operation.

Now, if any unwanted electrical interference manages to penetrate the outside shield, it will interact with both center conductors equally but with the same polarity.

The effect in the differential amplifier is that these same polarity voltages aren’t processed and effectively cancel each other out - the noise disappears.

This ability of balanced lines to reject noise and interference makes them popular when it is necessary to send signals over long distances.

This article republished with permission from Whirlwind.

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Posted by admin on 04/05 at 01:42 PM
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Tuesday, April 03, 2012

St Matthew Church Goes Fully Digital With DiGiCo Consoles

In 2009, St. Matthew United Methodist Church in Belleville, IL, undertook a major renovation of its technology systems, including making the leap from analog to digital by installing a DiGiCo SD8 as its main audio console.

Continuing its digital progression, the house of worship recently added an SD9 with D-Rack (connected to the SD8 via DiGiCo’s Little Red Box), replacing an analog desk at the hub of its video production suite to handle the increased complexity of its productions.

Church media consultant Phil Mahder of Training Resources and Ben Shipman, president of AVA Audio Video Associates again assisted in the transition, working with house broadcast engineer DJ Rockwell.

“St. Matthew has been pleased with their SD8 at FOH since they got it 2 years ago,” Mahder says. “The production level for their large dramas has made the mixer a must—even their Sunday services have had so much complexity that they are using the potential of the SD8 routinely. Not only do they have a lot of sources on stage, but they also do a rapid and complete changeover between services as they switch music styles. The church has been on local cable for many years, producing a quality product in both content and production value.

“Since getting the SD8, they have realized that the analog console in the video control room has been a limitation,” he continues. “Although they have remote control over the SD8 from the video station, there were conflicts at times between the goals of the FOH operators and the video operators. When the SD9 was introduced, we all quickly realized that this would be the perfect replacement for the analog console in the video control room, making FOH and video independent and completely digital. With the SD9’s affordable price tag, everyone was in agreement.”

The DiGiCo SD systems were Rockwell’s first foray into the digital mixing realm. Growing up at St. Matthew and actively involved in the media ministry for the past 10 years, he’s a third generation broadcast engineer with a keen interest in trending technology—particularly as it relates to video production. He found the DiGiCo desks to have a well-thought-out design offering an intuitive ease of use and fantastic sound quality.

“When we originally put the system in, back in 2009, we knew that there would be some more upgrades to come—especially for the video mix,” Rockwell says. “We started off mixing for video with an analog console, and then switched to a computer running the SD8 remotely. We were all in agreement that the new SD9 would be ideal for us, and were able to demo one prior to purchase.

“In fact, I was able to make a demo reel that showed how much it would improve the quality of our videos, which very much helped to convince the committees in charge of granting the funding. The SD9 proved to be the perfect solution for our problem and improved our productions greatly.”

Once the console was installed, Rockwell recalled, it was ready to run within an hour. “It sounded great and the processing was seamless. The snapshots were very smooth and easily customizable. St. Matthew is a house of worship that has been into technology for a long time, and we do our best to stay on the cutting edge. We started airing our services on cable more than 25 years ago, and now also stream live.

“On Sundays, we have three services back-to-back with no time for rehearsals in between and can be as simple as three mics and as extreme as 75-plus inputs. The SD8 and SD9 made these 15- and 20-minute switchovers—from Traditional to Contemporary to Blended with choir and orchestra—possible. Besides our normal worship services, we also put on two technically intensive productions each year, at Easter and Christmas, with a cast, crew, and orchestra of over 300. Having this kind of digital footprint—with all the bells and whistles it affords—is a must-have for what we do.”

DiGiCo’s Little Red Box (LRB) played an integral role in integrating the SD8 with the SD9. “It allowed us to take the second MADI I/O on the SD8, run it into the LRB—which is then sent to the SD9 over CAT5E. This enables us to send the local inputs on the SD8 to the SD9 using direct outs.”

From the crew to the congregation, everyone at the church has been extremely pleased with the SD9. “It’s really improved the audio quality of our worship services and productions,” Rockwell concludes. “We stream our services live and air them on cable, the quality of which has noticeably improved by our members who watch on line or at home.”

DiGiCo

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Posted by Keith Clark on 04/03 at 01:07 PM
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HHB Introduces Mogami 3306 Ethernet Cable For Mobile Applications

HHB Communications has announced the release of new Mogami 3306 Ethernet cable drums, specifically designed for demanding mobile applications.

Available in 25, 50 and 100 meter lengths, the new Mogami Ethernet cable is flexible enough to lay flat on a floor, yet rugged enough for reliable performance, even with the frequent set-ups needed in live sound and commercial venues. And it fully complies with TIA/EIA-568B Category 5e termination standards and performance characteristics.

The Mogami Ethernet cable is not a standard Cat 5e cable with a heavy-duty jacket; it is carefully constructed so that its four twisted pairs remain separated from each other, which helps to ensure consistent data rates, even under extreme conditions. Its durable construction also means that it can withstand being run over by a truck without loss of bandwidth.

Supplied on a professional Schill cable drum, the cables offer Neutrik RJ45 etherCON connectors finished with protective rubber caps.

“The 3306 is not an ordinary Cat 5e cable because it is custom designed for durability and consistent data bandwidth in the harshest of environments,” says HHB International sales manager Matthew Fletcher. “Live sound and broadcast engineers will rely on it for its durability in the field, systems integrators will be amazed by its consistent data rates, and its high-quality construction and heavy-duty jacket ensure that it lives up to Mogami’s reputation as a manufacturer of premium cables for audio professionals.”

HHB was awarded the exclusive distribution rights of Mogami bulk cables in the UK and Ireland last year and recently released a line of packaged cable products, which includes 46 premium cables to suit the needs of broadcasters, engineers and musicians. More specifics are available here.

HHB Communications

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Posted by Keith Clark on 04/03 at 09:57 AM
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Changing Times: Just What Is A/V Over IP?

An increasingly wide range of systems are being merged into the traditional IT network architecture
This article is provided by Corporate Tech Decisions

 
Before A/V and IT systems started merging together into one overall network, organizations typically managed each system separately.

Data signals were routed through IT’s servers and sent out to end users over Cat-5 cables, video traffic was contained within its own platform and ran over coaxial cable, and phone calls transited a private branch exchange (PBX) system before being carried to the desktop via an old school Cat-3 cable.

The systems were usually managed by different groups, with no crossover in equipment or expertise.

But today, those disparate systems are gradually coming together, and a single cabling backbone is often the launch pad for companies interested in converging their A/V and IP networks.

“I think what we’re seeing now is that the A/V industry probably has a more structured cabling approach, which very much mirrors where folks who’ve been exclusively IT-, data-, or telephony-focused in the past have already gone,” says Derek Joncas, manager of product marketing at Extron Electronics in Anaheim, CA.

An increasingly wide range of systems are being merged into the traditional IT network architecture, including Voice over IP telephony solutions, videoconferencing platforms and presentation systems.

And because conventional data cabling is ubiquitous in most modern buildings, a shared backbone is attractive to many organizations, who can often save money by using existing cables to distribute A/V signals throughout their facilities.

Misconceptions abound about what AV-IT convergence really is, says Ken Colson, vice president of sales and engineering at Tucker, GA-based LMI Systems. “A lot of people assume when you say A/V over IP, you’re simply running an audio/visual signal over a category cable, like Cat-5,” he says.

While that may indeed be the limit to convergence in some situations, other organizations have progressed to the implementation of more holistic network architectures, which often share switching equipment and other components in addition to backbone cabling. In those increasingly converged environments, the distribution of an A/V signal frequently occurs in a way that directly mirrors more conventional IP-only networks.

“A/V over IP is the ability to take analog or high-definition audio/visual signals and inject them into a network — either the existing IP network or it could be a closed network (meaning it’s separate from telephony or data traffic) — and distribute it to multiple endpoints,” Colson explains.

With the evolution of A/V and IT technologies, Joncas says the line between the two disciplines is blurring. “There’s not a big difference between how you manage a computer or server versus how you would manage an A/V appliance,” he says.

Those similarities are leading more organizations to merge their previously standalone A/V systems into their overall IP network architecture. “IT administrators may be a little more comfortable with the idea that you can have many more A/V appliances on your network nowadays, and have some confidence that you’re going to be able to manage them,” Colson says.

Addressing Delays

As A/V traffic increasingly moves from just sharing cables within the IT network to actually moving through some of the same switches and other hardware components, one potential issue administrators must be ready to address is network latency (or a delay in processing network data).

“When you think of an A/V network nowadays, a lot of the information that’s being exchanged is very, very high speed data that has a very low latency requirement,” Joncas says. “If you pair that with a traditional IT network, that latency requirement doesn’t disappear.”

He adds something as innocuous as users browsing the web could inject increased latency into the network, but adds that most of today’s A/V devices include the processing capabilities needed to help manage and overcome the potential latency and quality of service concerns that may crop up when layering A/V signals over an IP network.

Scenarios where users are consuming content without any reference to when the content was generated may have a greater tolerance for network latency, Joncas explains, but “when you’re dealing with live signals, latency is the most important factor.” He cautions that careful design of the network’s architecture is paramount to managing quality of service issues.

Proactively addressing network latency and bandwidth issues could involve adding or upgrading equipment or services on the existing IT network, Colson notes. “One of the challenges you have with the need to distribute A/V signals is some resistance from IT directors as far as putting what they consider to be a bandwidth hog on their network.”

Moving from a data-only environment to a mixed environment may also require that IT groups increase their knowledge of how A/V really works, he says, adding that basics such as “understanding how resolution needs — whether it be standard definition or high definition — equates to bandwidth requirements to push A/V through that network” are crucial to designing a network that can successfully support bandwidth-intensive, low-latency applications.

Colson says that organizations that rely on older networks may find it necessary to upgrade their switches to manage video priority, or even add switches or change to a virtual LAN to achieve the sort of traffic separation their particular case requires.

The convergence of A/V and IT infrastructures will look different in every enterprise. Each organization must carefully evaluate its needs, the level of funds they can devote to either developing a single robust architecture or multiple standalone systems, and the expertise available to them to manage a wide range of components within a holistic network or to instead oversee the provisioning of each platform individually.

Where those needs and resources come together will ultimately dictate where the various systems share resources and where they remain disparate.

For more articles like this, go to Corporate Tech Decisions.

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Posted by Keith Clark on 04/03 at 09:39 AM
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Monday, April 02, 2012

Low-Voltage Audio Products: The Series

This first article in a multi-part series discusses some of the challenges associated with using low-voltage audio information appliances.
This article is provided by Rane Corporation.

 
This is the first in a multi-part series. Additional segments are available here.

We live in an interesting age full of mind-numbing technical advancements and funny contradictions.

It’s ironic in this computer age with corporate predictors saying that low-voltage audio information appliances are the next big thing that a completely mechanical device consisting of a platform, a stick and two wheels was just as popular as one of the most sophisticated computers ever developed.

Another interesting contradiction is being able to obtain audio off the Internet from thousands of different sources, being able to store hours of full-bandwidth audio in a low-cost Web access device, with no moving parts, not much bigger than a package of gum, only to have it sound not much better than your best friend next door yelling at you through a tin can and waxed string.

Okay, it’s not that bad. But it can be a lot better.

Let’s look at the class of low-voltage audio devices called Information (or Internet, both are used interchangeably) appliances (IA), and let’s define them as anything connected to the Internet other than a PC that includes audio. Things like:

• Digital Audio Players
• Smart Phones
• Automotive & Home audio players
• Digital Cameras and Camcorders
• PDA ‘s
• Internet Radio and Cinema
• Internet Game Consoles

Speech-recognition is also included in, well, all of these.

As overpriced or uncompelling as some of these IAs are, successful technology is not far off. Along with everything else, achieving that success depends on good audio.

It cannot be left behind. If you doubt this, try listening to a home theater system without surround sound and a subwoofer. Good audio predicts good success.

Yet the audio quality on most devices is inferior. Typical signal-to-noise ratios measure in the 60 to 70 dB range (re -20 dBFS), which is about 5-10 dB worse noise than good portable CD players.

They tend to suffer from non-flat frequency responses (bass and treble boost being the biggest offenders) with limited low frequency response (typically rolled off beginning at 60 Hz apparently to compensate for lousy headphones) and some high frequency responses stopping as low as 5 kHz. (Line outputs have better frequency response and noise level than headphone outputs.)

Total harmonic distortion plus noise is not awful, usually below 0.1%. None of this is MP3’s fault. (See Karlheinz Branderburg’s excellent overview paper, “MP3 and AAC Explained” contained in The Proceedings of the AES 17th International Conference: High-Quality Bit Coding, Audio Engineering Society, 1999.)

Throughout all the early years, cassette machines sold their convenience over phonograph records and downplayed their inferior audio quality.

But eventually market competition brought the audio quality level up to par, or even better than phonographs. It seems we are repeating history once again with inferior audio quality on IAs. Hopefully the demand for higher performance will accelerate the process.

But it’s not going to be easy. Blesser & Pilkington in “Global Paradigm Shifts in the Audio Industry” (J. Audio Eng. Soc., Vol. 48, Nos. 9 and 10, September and October 2000) point out persuasively that consumers are more than willing to give up audio quality for convenience, portability and price.

Their report uses the rapid growth in Internet audio as an example of the acceptance of low-quality audio by consumers willing to make the trade-off for something totally new (MP3 audio, for instance).

In spite of what is claimed, there is no comparison between Internet audio and CD-quality, yet the success of MP3 is undeniable—overwhelming even. It is hard to find any examples that compare to this phenomenon. And it happened with audio quality that is “good enough.”

This lets us know in no uncertain terms that the consumer side of the audio business is running things, not the pro audio side. DVD-A, for example, is never going to impact the world the way that MP3 already has.

While the committees fought and scrapped over bit-count and higher sampling rates, the MP3 folks were out changing the world. Lesson learned—but that doesn’t mean we have to accept the quality. We can help make it better.

It is more than a little ironic that with many information appliances and other low-voltage audio products it is the analog parts and circuitry that are degrading the audio.

As will be seen below, the latest audio converters are truly impressive. Based on the best delta-sigma technology, using extreme oversampling and advanced noise shaping, they leave little room for criticism.

Good audio requires good hardware. Important issues include interconnection, grounding, testing, noise, and selecting good parts.

Too many designers think their job is done once they deliver the decoded data stream to the D/A converter on one end, or figure the A/D converter on the input end solves all their analog needs.

A complete step-by-step, nitty-gritty, smoke and mirrors design process for achieving high-quality low-voltage audio circuits lies outside the scope of this paper.

However, the important issues are illuminated with tips, pointers and Internet Web addresses for obtaining additional information. If you have audio circuits to design, what follows should help.

Interconnection
A full-featured information appliance usually contains several digital interfaces with digital audio capabilities.

USB streaming controllers allow digital audio systems to connect bidirectionally to host systems via the USB port used in almost all PCs and Macs.

In addition, many IAs support the S/PDIF digital interfacing standard used in DVD, CD and MD players. Likewise, Firewire (IEEE-1394) is emerging as the preferred digital interconnect for consumer electronics, computer peripherals, and professional audio/video networks.

Direct analog audio interfaces need careful attention to input and output stage design. Balanced, or differential, inputs and outputs are the best choice, offering noise immunity and greater dynamic range, but obviously not all of the real world agrees with the best choice.

Designers must accommodate all sorts of unbalanced accessories and equipment—everything from stereo headphones to PCs.

Treat unbalanced signals as floating or quasi-balanced sources by connecting the signal and the return/ground/shield as a balanced pair into a difference or instrumentation amplifier (e.g., hook the hot wire to pin 2 and the return wire to pin 3 of an XLR connector).

Float the outputs and drive them differentially or balanced. Use a high-quality differential output line amplifier to drive the signal line and return leg. When interconnecting to other equipment use carefully proven wiring techniques.

Interconnecting impedances must follow standard conventions of outputs low and inputs high. Keep output impedances in the 50-300 ohms range.

If an output amplifier requires more than this to remain stable when driving capacitive loads (i.e., long lines) then get rid of it and get something better—something inherently more stable and designed to drive long lines. Make line-level input impedances at least 10k ohm and mic input impedances lower, but not less than 1k ohm.

Grounding
Too often a connection between an unbalanced output and a balanced input produces hum and buzz. Consult Grounding and Shielding of Audio Devices and the special grounding issue of the AES Journal (J. Audio Eng. Soc., Vol. 43, June 1995) for detailed theory and application tips.

If inputs and outputs are kept balanced, or fully differential, with all cable shields bonded to the chassis immediately at the point of entry or exit, few grounding noise problems occur.

However, we live in the real world where unbalanced interconnects are the norm and cable shields are too often grounded to the signal reference lines or at a distance too far from the input/output to be effective.

As discussed below, this is another good reason to design your audio system fully differential. Fully differential lines are immune from ground-induced noises. It is with the conversion from differential to single-ended that problems begin.

It can get so severe that special jerry-rigged cable assemblies are proposed to make interfacing different equipment easier (Brown, Pat, “Universal Interface Cable?” Syn-Aud-Con Newsletter, Vol. 29, pp. 18-19, Winter 2001).

This suggested interconnecting cable allows the four possibilities of connecting the shield to be quickly tried: tied to both chassis; tied only to the sending end; tied only to the receiving end; tied to pin 1 of the receiving device.
Testing & Documenting the Results

Judging from the lack of audio specifications on IA data sheets and sales literature, it is tempting to conclude that manufacturers do not print audio specs because they have never measured them.

It is shocking, for instance, to overhear a conversation by a manufacturer of an information appliance with audio (in fact, audio is its only mission) and to hear him say, “What’s that?” when asked about Audio Precision measurements. But, to be fair, the lack of audio specs probably has more to do with the customers not asking for them than anything else.

From an audio quality standpoint, objectively comparing IA products is impossible without full specifications. Missing on most data sheets are basic audio specifications—SNR, THD+N, and frequency response—the basics.

If found at all, the data is meaningless since test conditions are not stated. Audio specifications come with conditions. Tests are not performed in a vacuum with random parameters. They are conducted using rigorous procedures and the conditions must be stated along with the test results.

Many designers of Internet information appliances are new to audio and don’t know where to turn to learn about audio testing and how to state specifications. Here are some resources:

• Basic analog audio tests are well covered by Metzler’s fine book on fundamentals (Metzler, R.E. Audio Measurement Handbook (Audio Precision Inc, Beaverton, OR 1993). See Audio Specifications for preferred testing conditions.
• Testing digital audio products is covered in Cabot’s AES tutorial paper (Cabot, Richard C. “Fundamentals of Modern Audio Measurement,” J. Audio Eng. Soc., Vol. 47, pp. 738-762, Sep., 1999). Fundamental concepts in analog and digital testing of audio equipment are reviewed, including tradeoffs inherent in the various approaches, the technologies used, and their limitations.
• The unique requirements of testing all digital audio power amplifiers is covered in Neesgaard’s application note, available from Texas Instruments.

Accurate audio measurements are difficult and expensive. The test equipment necessary to perform all the common audio tests costs a minimum of $10,000.

And that price is for computer-controlled analog test equipment, if you step-up into the digital-based, dual domain stuff—double it. This is why virtually all purchasers of IAs must rely on the honesty and integrity of the manufacturers involved, and the accuracy and completeness of their data sheets and sales materials.

Stay tuned for the coming articles in this series. Want to get a jump on the reading? Head on over to the Rane Website where you can read this article in its entirety.

Supplied by Rane. For more, go to rane.com

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Posted by admin on 04/02 at 02:33 PM
AVFeaturePollAVDigitalInstallationInterconnectSignalSound ReinforcementTechnicianAudioPermalink

Extron Announces Two Output HDMI Distribution Amplifier

Extron Electronics has introduced the HDMI DA2, a one-input, two-output distribution amplifier for HDMI video and embedded multi-channel digital audio.

The HDMI DA2 supports HDMI specification features including data rates up to 6.75 Gbps, Deep Color up to 12-bit, 3D, Lip Sync, and HD lossless audio formats.

This HDCP-compliant distribution amplifier supports all HDTV rates including 1080p/60 and PC resolutions up to 1920x1200. It features two Extron-exclusive technologies: EDID Minder, which maintains continuous EDID communication between connected devices; and Key Minder, which authenticates and maintains continuous HDCP encryption between input and output devices.

The compact HDMI DA2 is well-suited for applications that require the distribution of an HDMI source signal to two displays.

“AV system designers and integrators have been asking for a distribution amplifier that not only splits an HDMI signal but manages communication between the source and the displays,” says Casey Hall, vice president of sales and marketing for Extron. “The latest addition to our growing line of HDMI products, the HDMI DA2 with EDID Minder, fills their need for a reliable, high performance HDMI distribution amplifier, and offers a host of integrator-friendly features.”

To enhance and simplify integration, the HDMI DA2 offers integrator-friendly features, including automatic input cable equalization, automatic bit depth management, selectable output muting, and indicators for monitoring and troubleshooting. Input cable equalization restores and reshapes incoming HDMI signals, reducing the need for additional signal conditioning equipment by compensating for weak source signals or signal loss from a long input cable.

The HDMI DA2 automatically adjusts color bit depth based on the display EDID, preventing color compatibility conflicts between source and display. Outputs can be muted independently via RS-232, allowing content to be previewed on a local monitor. Additionally, the distribution amplifier provides immediate visual confirmation of EDID status, HDCP authentication, and signal presence confirmation for each port via front panel LED indicators.

Extron Electronics

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Posted by Keith Clark on 04/02 at 01:50 PM
AVLive SoundNewsPollProductAVInstallationInterconnectMonitoringRemoteSignalAudioPermalink

Reidel Introduces Suite Of AVB products For Artist Digital Matrix Intercom Platform

Reidel Communications has premiered a suite of AVB products for the Artist digital matrix intercom platform. AVB allows for transporting AES3/EBU audio in real-time with guaranteed bandwidth and quality of service (QoS) via Ethernet-based Local Area Networks (LAN). The new products were unveiled at the recent Prolight+Sound 2012 show in Frankfurt.

Based on official IEEE next generation Ethernet standards like 802.1Qav, P802.1Qat and P802.1AS, AVB allows risk-free utilization of AVB compliant facility or enterprise LAN infrastructure for intercom applications. This allows for new approaches in system and facility design providing significant savings in infrastructure investments.

Intercom applications for Riedel’s AVB products feature matrix-to-control panel connections via LAN, audio distribution via LAN, matrix-to-matrix trunking connections via LAN and distribution of digital partylines via LAN.

The Riedel suite of AVB products includes the AVB-108 G2 Client Card as well as the Connect AVBx8 panel interface.

The AVB-108 G2 card is a regular Artist client card to be used inside the Artist mainframe. It converts eight Artist matrix ports into AVB and vice versa.

The AVB-108 G2 client card communicates either with other AVB-108 G2 client cards in another Artist systems, e.g. for trunking, or with Riedel’s Connect AVBx8 panel interface.

The Connect AVB Panel Interface is a small unit, which converts an AES signal into AVB and

Connect AVBx8 converts eight AES signals to AVB and vice versa. Build in a compact 9.5”/1RU housing the device provides eight CAT5 ports to connect up to eight Artist control panels in one or two-channel mode to the matrix via IP-based LANs. Connect AVBx8 is the perfect team mate for Riedel’s AVB-108 G2 eight channel AVB client card.

Reidel Communications

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Posted by Keith Clark on 04/02 at 01:39 PM
AVLive SoundNewsPollProductAVDigitalInterconnectNetworkingSignalAudioPermalink

Thirteen Audinate Dante-Enabled Products Introduced At Prolight+Sound 2012

At the recent Prolight+Sound 2012 show in Frankfurt, eight Audinate OEM partners announced a total of 13 Dante-enabled products.

“What is truly amazing is the wide range of different Dante products that were launched at the show”, says Lee Ellison, Audinate CEO. “It seems that no matter what you are looking for, there is now a Dante-enabled product on the market.” Lee also adds, “The market is dictating what it wants, and Dante seems to be the answer,”

Here is what was launched at this year’s show:

—Yamaha Commercial Audio Systems announced the launch of the new CL Series of digital consoles. The Yamaha CL Series is a Dante network–based console featuring remote I/O for a faster, more responsive Yamaha system solution. All three CL models in the Centralogic series, only differentiated by frame size and input capability, feature 24 mix buses, 8 matrix buses, plus stereo and mono outputs, and 16 DCAs. The footprint of all three CL consoles is small, yet powerful and has been developed specifically for sound reinforcement applications such as performing arts venues, theaters, houses of worship, touring, and remote broadcast.

—Symetrix announced the SymNet Edge, a significant update to its long line of DSP solutions packed with new features. Symetrix designed Edge to meet the I/O capacity requirements of the bulk of commercial sound installations. Edge features four configurable I/O card slots, up to 16 channels total of local I/O plus 128 (64x64) channels of Audinate’s award winning Dante network audio.

—Harman Soundcraft is extending its range of digital audio transport option cards for its Vi, Si Compact and Si1/2/3 Series digital consoles with all options expected to start shipping this year. Plans are also in place for Dante network cards compatible with Vi, Si Compact and Si1/2/3 Series consoles, following an agreement reached between Soundcraft and Audinate. “This agreement is a reflection of Harman’s responsiveness to its customers who want Dante networking,” adds Ellison.

—NEXO announced the new NXDT104 Dante audio plug-in card for the NXAMP, enabling NEXO loudspeaker systems to explore the many benefits of a high-performance AVB-ready digital networking solution. The NXDT104 will distribute digital audio plus integrated control data, automatically configuring its network interface and finding other Dante devices on the network.

—AuviTran, the networking specialists, is adding Dante technology to the Audio Toolbox product range. Audinate was chosen to accelerate AuviTran’s development towards new networking technologies. Since 2003 AuviTran has provided audio professionals with innovative networking solutions using EtherSound technology and now adds Audinate’s Dante networking to its networking solution portfolio.
 
—JoeCo Limited introduced the latest product in its BlackBox range of multi-channel live audio recorders and players in the form of the new BLACKBOX BBR64-DANTE RECORDER that can record or replay 64 channels.

—Four Audio, manufacturer of audio and loudspeaker management systems, announced the launch of the new DBO1, a Dante-enabled breakout box with 8 analog output channels. It allows the user to go from Dante to analog signals in a small 1U box. The DBO1 is operated by simply connecting it to a Dante network and configuring the easy-to-use breakout box via the Dante Controller.

—TEQSAS, a professional audio solutions company with core competencies in delivering high quality audio products, announced a new Dante AES breakout module for the cyberTEQ - m-family of digital audio interconnect solutions.

Audinate

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Posted by Keith Clark on 04/02 at 08:48 AM
AVLive SoundNewsPollProductAVBusinessDigitalEthernetInterconnectNetworkingSignalAudioPermalink

Friday, March 30, 2012

Tour De Absurd: Unbound By The Fundamental Rules Of Reality

Everybody's dealt with horrible vendors from time to time, and Sully's got some tales from the road to which we all can relate.

I was just bitten by a dog.

Truthfully, not 20 minutes ago when I went to pick up a piece of gear at somebody’s house.

When I pulled into the driveway, an unholy spawn of a late night dalliance between Benji and the Geico gecko waddled over to me, growled, then bit me on the f***ing ankle.

I screamed like a five year old, which somehow triggered the garage door to open and spew a teenage girl carrying the gear I was there for.

“This yours?” she piped. “Your dog just bit me on the f***ing ankle,” I squeaked. “”Really? Sorry…” She froze with a look on her face that indicated she was now invisible and I should leave wondering where’d that girl go?

Needing more satisfaction I called the owner of the house.

“Hello?”

“Your dog just bit me on the f***ing ankle.”

“That dog’s 13 years old, he’s never bitten anyone.”

“Oh. Cool. Never mind then.”

“You sure?”

“Hang on a tic, let me make sure my portable morphine drip isn’t on high. Nope, machine’s good… the f***er definitely bit me”

The vicious dog attack left me sulking about the hound’s total lack of fear and respect for me. Then I got mad at myself for sulking about not being feared by an arthritic Chihuahua. Skillfully, I managed to cram in a 30-minute session of bi-polar self-loathing and admonishment in the time it took to drive from the scene of the assault to our bus.

It suddenly occurred to me, as I stared down at the dog sticking out of my jeans, that this was a fitting coda to the four-week tour de absurd that I and the rest of my crew had just endured. During the preceding month, 90 percent of the production vendors we had met had attempted to convince us they alone were not bound by the fundamental rules of reality.

To prove this point, they had taken our advance phone calls, listened carefully to our requests, sagely reassured us all would be well… then rolled us over and tried to bite us on the neck when we showed up. Same deal as the dog. They looked us up and down and figured they could take us.

Act 1
Me: “Hey, how wide is this box?”

PA prestidigitator: “205 degrees for the long throw, 365 degrees for the downfill.”

Me: (Knowing it’s general admission) “OK.”

 
Act 2
The setting: a large field with bands of disgruntled raisins milling arrogantly about.

A2: “We’re ready, my lord. We are prepared for you to communicate with the magic box and give us the array angles for the sound system.”

Steak sauce: “I have spoken with the machine. It gives no advice today. You must have done something to anger it. Go now, butcher the factory program and burn the fatted DSP as an offering. Leave me.”

A2: “My liege, the troubadours will be upon us soon… Can you offer no wisdom for us to assuage their FOH knight?”

Steak Sauce: “Tell him…tell him… the sound will emerge crooked if you angle the speakers. Tell him flat… Yes, flat is best. Threaten to rub petroleum jelly on him and burn him as a witch if he questions you.”

A2: “You are indeed the wisest in the land.”

 
Act 3
Production manager motioning to four speakers hanging from swing chain flown with the aid of two winches off the front of a quad runner.

PM: “What da hell is that?”

Local vendor: “EV X-Array.”

PM: “No it’s not.”

Local vendor: “Yes it is.”

PM: “No it’s not.”

Local vendor: “Yes it is.”

PM: “No it’s not.”

Local vendor: “O.K., no it’s not. I bought an X-Array box and copied it.”

PM: “You mean EV X-Line. You copied X-Line.”

Local vendor: “Yeah, the big EV box. X-Array-Line.”

PM: “O.K., just so we’re clear…you pirated something from EV and call it X-Array.”

Local vendor: “Yeah. Sounds great too. Wanna see our V-Disc wedges?”

Act 4
The three principal characters enter upstage center and proceed downstage in slow motion, their movements reminiscent of Apollo astronauts bravely approaching an ill-fated capsule.

Bonded by an invisible energy, their gaze begins tracking the seventy-five degree seating angle until at last their eyes settle upon the top seat, 600 feet aloft. One holds a laser range finder and whistles quietly at the data it yields.

Their attention is suddenly diverted to the single horizontal row of two EAW KF750s stacked neatly on the stage deck. A small man rapidly approaches the group.

He is equipped with a large black belt dubiously supporting a brick-like walkie-talkie with a solid three-foot antenna fully extended.

The effect is not unlike a remotely controlled Hobbit. A roll of gray tape used to seal air conditioning vents dangles from his meaty wrist, and he is thrusting an irate digit at the tiny speaker array.

Small Man With Big Belt: “I don’t want to hear it! Them speakers cover front row to top row perfect. They’re 70 degrees up and down so we don’t even need to tilt them. Sounds exactly up there like it do down here. I don’t want any of your smart-alecky talk about math. We done it this way for 10 years and it sounds great. Now, welcome and go away, I mix the opener tonight and I gotta make sure they’re happy”.

 
Act 5
A man stands beaten, his feet loosely clutching the prefabricated stage. His attention is captivated by the scene unfolding before his weary blue intelligent eyes…Men of ill-advised employment are hoisting a large-format console by attaching a 1/4-ton drape motor to its top-riveted session handles.

They stand under it, marveling at the graceful way it swings in the cool breeze. Our hero calculates that when the first handle lets go, the desk will swing low, hijack a stagehand at it’s nadir and force him to ride it bareback halfway to the rafters.

As the console reaches it’s apex and the second handle shears away, the desk will immediately divest itself of it’s passenger and enter a vertical spin, 25 feet off the ground, shortly proving wrong the load-out adage, “gravity is your friend”.

Quickly, without remorse, the sad man dispatches an intern to the balcony with a bin of economy popcorn and two video cameras. Word must reach the outside world of the transgressions that have transpired here…

 
Act 6
Me: “What version of the prediction software are you using?”

Them: “Ashly crossovers. They’re out front.”

I own a cat that hides behind the drapes when in trouble. It sits perfectly still, avoiding all eye contact, staring straight ahead looking like a paisley tumor respirating below the front window.

She is so convinced of her sudden undetectability that I have no choice but to accept the fact that the curtains have spontaneously evolved a tail and I should look elsewhere for her.

I marvel at her ability to gaze directly into the face of truth and maintain plausible deniability. Like the vicious miniature wolfhound noted earlier, the cat has eyed me up and come to the conclusion that she’s got my number.

I’d start dutifully working on a complex about my lack of respectability within the various animal phyla, but I know from experience, it’s not just me.

Many of the band guys I run into step off of the bus in the morning with dingoes latched to their ankles. They all have stories that somehow involve PA and lighting vendors avoiding eye contact and hiding behind backdrops with only their five D-MAG lights sticking out.

Sometimes I’ll look into their eyes, pat their dogs and smile with them, offering these words of solace: “get your sun block out boys, we’re goin’ to Hell.”

 
Finale
Me: “Two horns are popping red and two are green. Which is correct?”

System provider: “Which is better?”

 
Sully is a veteran live sound engineer and really has no clever off-hand remarks for this space at this time.

{extended}
Posted by admin on 03/30 at 10:54 AM
Live SoundFeaturePollSlideshowEducationEngineerInterconnectLine ArrayLoudspeakerSignalSound ReinforcementSystemAudioPermalink

Advanced Small Console Techniques: Maximizing The Available Feature Set

What about those times when, for whatever reason, a big console is not available? That's the time for ingenuity and some special techniques

Everyone loves a big console.

Even when they’re small in size, like modern digital consoles, we favor consoles that have everything we need to solve any problem that may come up.

But what about those times when, for whatever reason, a big console is not available?

That’s the time for ingenuity and some special techniques that maximize the usefulness of the available feature set. Some people call these “workarounds,” but the term I like is “tricks.”

You know, like a magician. Here are some of my favorite tricks.

The XLR “Y” Cable
The one-female-to-two-male XLR “Y” cable is a powerful problem solver that should be in every audio person’s bag of tricks. It’s most common use is splitting an input to two channels of a mixer, using one channel monitors, the other for mains. This allows the EQ and processing used on one to not appear in the other.

It can be especially important if you need maximum gain before feedback in the monitors for a vocalist, or the input channel EQ helps with an acoustic guitar that’s giving you feedback issues. This trick can also be used for “wet/dry” channels, or if you need some sort of crazy EQ or effect for one part of a song and then quickly need to change back to normal.

Channel Insert As FX Loop
When a console is lacking a function, sometimes it can be found in a piece of outboard gear. For example, most effects units have a built-in mixing feature. When the aux sends are all used up, but you still want one more special effect on one more input, it’s a simple thing to insert the device into the input channel, and use the effect’s wet/dry mix control to vary the amount of effect.

No, it’s not ideal, but under battle conditions, you do what you must.

Direct Out As FX Send
The input channel’s direct output is not just for multi-tracking any more! An input channel’s direct output is another way to get signal to an effect that is only needed on one channel. This trick works well for things like a chorus on an acoustic guitar, or a reverb on the lead vocalist.

It’s best if the direct out is post-fader. A pre-fader can also be used as a direct out too, but you’ll have to keep a watchful eye (ear) on relative levels.

Audio Group As FX Send
So, you’re using four auxes for monitors and two for vocal effects - how are you going to get reverb on the drums?

Sure, the same ‘verb could be used for everything, but where’s the fun in that?

One trick is to assign the drums you’d like to have reverb on (I often leave out the kick drum) to a sub-group, and use the sub-group output as an effects send.

If you have a 2-in/2-out effects box, assign the drums to two sub-groups and you have stereo drums with stereo effects. Use the group’s insert loop and the mixing feature of the effects box, and you have it without the need for an effects send or return.

Pretty cool, huh? This trick can also be applied to vocals, horn sections, or any group of inputs.

Uncommon Insert Hardware
Everyone is familiar with dynamics processors and equalizers inserted into input channels. However, these are not the only useful tools you can insert into an input channel.

For example, many smaller mixers do not include a variable high-pass filter (HPF). The advent of digital loudspeaker processors finds a great many analog crossovers sitting around gathering dust. Just connect an insert send to the input of an analog crossover, then take the output from the crossover’s “high” output, and there you have it, a variable HPF.

Many common analog crossovers have a frequency range both above and below the frequency of a console’s switched HPF, which is usually 80 Hz to 100 Hz. If you don’t have an analog crossover laying around, the glut of these kinds of units on the used market makes them very affordable.

That 4-channel, 2-way crossover pulled from the old monitor system can now be four channels of variable HPF, all in a single rack space.

Sub-Mixers
Any conversation about small mixers should include sub-mixers.

Sub-mixers get a bad rap, but they are an excellent solution to many small mixer problems, the most common being not enough input channels.

The logical use for sub-mixers is on things that can be grouped together - things like drum kits, horn sections, or backing vocals.

With a stereo sub-mixer, you can pan some inputs left, some others right, effectively giving you two sub-mixers. Sending the sub-mixers outputs to input channels on the main mixer transforms those input channels into additional sub-groups.

If your main mixer has sub-groups, you can also patch in your sub mixer there, and save the inputs on the main mixer for other things.

Insert To Bypass Mic Preamp
It’s becoming increasingly popular for artists or their engineers to carry around an esoteric front-end device. These devices almost always contain a microphone preamp as well as some combination of EQ and dynamics processing.

More often than not, the line inputs of an inexpensive mixer is the mic input padded down to line level. This means the fancy preamp is being hooked up to the console preamp, which is one preamp too many.

A specially constructed tip-ring-sleeve cable can bypass the mixer preamp completely by using the channel insert jack. The preamp end of the cable is wired tip/sleeve, the mixer end of the cable is wired ring/sleeve.

Please note that a cable constructed this way is for use with the most common unbalanced inserts that are tip send/ring return. For a mixer that’s configured ring send/tip return, both ends of the cable should have a tip/ring/sleeve connector wired tip/sleeve.

These cables, and any other cable wired in a non-standard fashion, should be very well labeled!

Many Happy Returns
Console returns are very often left unused. Most of us prefer the stereo input channels and/or the mic input channels, and for good reason. They have EQ and full routing capability, so what’s not to like?

Once again, for things like drum kit reverb where you are going for a “room vibe” type of effect that will be set and mostly left alone, the often overlooked console return works just fine. These are good places to plug in sub-mixers too.

Go Forth And Mix
These tricks are ways to squeeze big console performance out of a small console (or two). Again, some of them are not preferred practice, but they allow you to say “sure, no problem” instead of “I can’t do that.”

Now go forth and mix - big console or small.

Dave Dermont is a long-time working live sound mixer and a moderator of the ProSoundWeb Live Audio Board (LAB).

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Posted by Keith Clark on 03/30 at 10:25 AM
Live SoundFeaturePollAnalogConsolesDigitalEngineerInterconnectProcessorTechnicianPermalink

Thursday, March 22, 2012

Universal Audio Now Shipping Apollo Audio Interface With Real-Time UAD Processing

Get more of the latest news from the 2012 PL+S show.

 
Universal Audio (UA) has announced the worldwide shipment of Apollo, a new high-resolution audio interface that combines UA’s analog design heritage with UAD powered plug-Ins in a sleek, elegant recording system for Mac and PC.

“Apollo is the culmination of 10 years of analog and digital audio development here at UA,” says Bill Putnam Jr., Universal Audio founder. “In many ways, it’s brought the analog and digital sides of our company together. With Apollo, we’re delivering the sound, feel, and flow of analog recording with all the conveniences of modern digital equipment, including next-generation Thunderbolt technology.”

Coinciding with initial product shipments, Apollo is also on display at the ongoing Prolight+Sound 2012 show (booth C68 in hall 5.1), with live demonstrations by Fab Dupont (Jennifer Lopez, Mark Ronson, Santigold) and vocalist Liza Colby.

Apollo is now available in both DUO CORE and QUAD CORE processing formats with estimated street prices of $1,999 (DUO processing model) and $2,499 (QUAD processing model).

In addition, Apollo’s Thunderbolt Option Card will be shipping in summer 2012, with pricing TBD.

Note that Apollo is compatible with Mac OS X 10.6 and 10.7; Windows 7 support coming in summer 2012.


image

   

Universal Audio

 

{extended}
Posted by Keith Clark on 03/22 at 07:14 AM
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Wednesday, March 21, 2012

Connecting Unbalanced Outputs To Balanced Inputs—And Vice-Versa

Two key issues: different signal operating levels between consumer and professional equipment, and making connections while avoiding ground loop problems

Based on my years of helping customers solve interfacing problems of all sorts, connecting unbalanced outputs to balanced inputs, and vice-versa, certainly ranks among the most common and confusing of tasks for system integrators.

Basically, two issues must be dealt with. The first involves the different signal operating levels between unbalanced (consumer) and balanced (professional) equipment. The second involves making the actual connections to transfer the signal while avoiding “ground loop” noise problems.

Signal operating and reference levels are significantly different for consumer and professional equipment. The consumer reference level is -10 dBV or 316 mV rms, while the pro reference is +4 dBu or 1.228 V rrns. Therefore, a voltage gain (for consumer outputs driving pro inputs) or loss (for pro outputs driving consumer inputs) of 3.9 or about 12 dB is theoretically required.

On the consumer to pro side, a fair question might be “Why not use a step-up transformer for this gain?” Several commercial products do, but I don’t recommend them. Let me explain. Transformers simply reflect impedances from one winding to another - they do not have an intrinsic impedance of their own. Assume we use a transformer with a turns (voltage) ratio of 1:4 to get 12 dB of gain. This unavoidably (laws of physics) makes the transformer’s impedance ratio 1:16, the square of its turns ratio.

Therefore, the impedance of the pro input will be reflected back to the consumer output as 1/16 of that. Since a typical balanced Input has an impedance somewhere between 10 kiloohms (kohms) and 40 kohms, it will be seen by the driving consumer output as 625 ohms to 2.5 kohms. Virtually all consumer outputs are rated to drive a “10 kohms minimum load.” That’s because their internal or “output” impedance (usually unspecified) is typically 1 kohm or more.

Therefore, actual gain will not be 12 dB but only 3 to 8 dB because of the low load impedance on the consumer output. Worse yet, the consumer output will experience a serious headroom loss, up to 8 dB, causing premature clipping. Since most consumer outputs use coupling capacitors designed for a “10-kohm minimum load,” the severe loading will usually result in poor bass response, too.

Usually, specs relating to these issues are conspicuously absent from manufacturers’ data sheets. However, gain is seldom an important issue because pro equipment inputs generally have at least 12 dB of additional gain “reach.” If we eliminate the signal gain requirement, unbalanced to balanced interfaces become fairly straightforward,

On the pro to consumer side, operating level differences are an important concern.

Because consumer inputs rarely include input level controls, the consumer equipment is easily overloaded by pro signal levels.

Again, since the professional reference is +4 dBu or 1.228 V rms and the consumer reference is -10 dBV or 316 mV rms, a loss of about 12 dB is required.

Obviously, the output of the pro equipment could be reduced by 12 dB, but then its level metering would be nearly useless and signal-to-noise performance would be degraded.

Signal attenuation is required for these interfaces.

In most cases, noise rejection is a far more important issue.

For consumer to pro interfaces, the widely used hookup of Figure 1 uses shielded single-conductor cable and an RCA to XLR adapter or ready-made adapter cables built as shown.

Figure 1 - Using an unbalanced cable with an adapter results in zero noise rejection.

Unfortunately, it has 0 dB of ground noise rejection - and wastes all the potential noise rejection of the balanced input! Sadly, the availability of such adapters or cables leads many unwary users to create this noise-prone connection. Performance is especially poor when cables are long, since the entire interface is unbalanced, allowing both audio and ground noise to flow in the cable shield.

A far better hookup shown in Figure 2 uses shielded twisted-pair cable to take advantage of the noise rejection available from the balanced input stage.

Figvre 2 - Using balanced cable wired as shown results in at least 30 dB rejection.

Because ground noise now flows in the shield conductor rather than one of the signal conductors, noise rejection is improved by about 30 dB when the input is a typical “active” differential-amplifier type. If the equipment’s balanced input is truly high-performance, using an input transformer or the InGenius IC, rejection is improved by about 80 dB. [Reference 1]

Figure 3 shows noise rejection for various consumer to pro interfaces. The top plot at 0 dB represents the simple adapter and 2-conductor cable connection.

Figure 3 - Noise rejection in unbalanced to balanced interface. Top to bottom: cable of Figure 1, cable of Figure 2, cable of Figure 2 plus output transformer, cable of Figure 2 plus input transformer.

The plot at -30 dB shows the improvement due to the simple 3-conductor hookup. The next plot shows the effect of an ordinary isolator using an output transformer (no internal Faraday shield!). It improves 60 Hz hum by about 20 dB, but has little effect on buzz artifacts around 3 kHz. A high-quality isolator using an input transformer (with an internal Faraday shield) increases rejection to almost 100 dB at 60 Hz and about 65 dB at 3 kHz.

For the best possible noise rejection, use a 3-conductor cable (wired as in Figure 2) from the unbalanced output to the isolator input. The plots here were measured with a 600-ohm unbalanced output and a 40-kohm balanced input.

For pro to consumer interfaces, rejection of ground noise is also very desirable.

Figure 4 (below) shows noise rejection for various balanced to unbalanced interfaces, The upper plot at 0 dB represents a direct connection, such as for an adapter or adapter cable.

Direct connections are problematic because various types of balanced output circuits are used in equipment each with its own peculiar limitations.

Some, such as the one in the schematic, can be damaged if one of its output terminals is grounded. Outputs stages using either transformers or widely used “servo-balanced” outputs, must have one terminal grounded in order to produce a proper output signal at the other. But the “servo-balanced” output can oscillate or become unstable if the ground connection is made at the far (receive) end of a cable. [Reference 2]

Figure 4 - Noise rejecfion in balanced to unbalanced interface. Top to bottom: direct wiring, connection with output transformer, connection with input transformer as in Figure 5.

Therefore, a cable that works with one piece of equipment may not work with another. Fortunately, adding a transformer allows the interface to work with any output stage. The middle plot shows that an output transformer reduces 60 Hz hum by about 50 dB and buzz artifacts around 3 kHz by about 20 dB. A high-quality input transformer, increases rejection to over 105 dB at 60 Hz and to nearly 75 dB at 3 kHz.

When this transformer is a 4:1 step-down type, the 12 dB level difference problem is neatly solved as well. Figure 5 shows the system schematic of this “universal” pro to consumer interface.

Figure 5 - A step-down transformer works with any balaoced output.

Bill Whitlock has served as president and chief engineer at Jensen Transformers for more than 15 years and is recognized as one of the foremost technical writers in professional audio.

REFERENCES
[1] Whitlock, B., Interconnection of Balanced and Unbalanced Equipment, Application Note AN003, Jensen Transformers, Inc., 1995.
[2] Hay, T, Differential Technology in Recording Consoles and the Impact of Transformerless Circuitry on Grounding Techniques; Audio Engineering Society, 67th Convention,1986, Preprint #1723.

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Posted by Keith Clark on 03/21 at 05:33 PM
AVFeaturePollStudy HallAVInterconnectPowerSignalAudioPermalink

Yamaha Launches New CL Series Digital Consoles At Prolight + Sound 2012

Get more of the latest news from the 2012 PL+S show.

 
Retaining essential features and functionality that have become standards over the past quarter of a century, the new Yamaha CL Series of digital mixing consoles offers an evolved experience in accessible mixing. The CL Series was just unveiled at the ongoing Prolight+Sound 2012 show in Frankfurt.

The line-up comprises three consoles, the CL1, CL3 and CL5, ranging in scale from 48 to 72 mono plus 8 stereo inputs. All feature 16 DCAs and 24 mix/8matrix output buses.

Founded on the proven Centralogic interface, the CL Series incorporates multiple innovations and refinements, including enhanced Select Channel functions and User Defined knobs. 

The CL Series EQ and effects have been vastly expanded. They include an Effect Rack with VCM analog circuitry modelling technology, as well as a Premium Rack that includes the Rupert Neve Designs Portico 5033 equalizer and Portico 5043 compressor/limiter, developed in close cooperation with Rupert Neve.

Two new rack-mountable I/O units, Rio3224-D and Rio1608-D, can be used in a variety of combinations and configurations, communicating via a scalable Audinate Dante digital audio network.

Up to eight I/O rack units can be connected to a CL Series console, while multiple CL consoles can share control of the same I/O rack unit. A new Gain Compensation function adds the ability to combine front of house and monitor control via a single network, for comprehensive digital live sound integration.

For live multitrack recording and virtual sound checking, CL consoles are equipped with dedicated recording control capabilities for use with Steinberg Nuendo Live DAW application running on a Windows or Mac computer.

The new Yamaha CL Series line-up. (click to enlarge)


Further Details

User interface. The Centralogic user interface ensures that the consoles will be immediately familiar to many thousands of live sound engineers.

It has evolved considerably on the new consoles, incorporating a new generation, highly responsive color touch screen and an array of user definable rotary encoders and buttons.

Newly designed faders offer optimum feel, visibility and accuracy and are freely configurable to allow control of any combination of inputs, outputs or the 16 DCA faders.

The control surface also provides editable, back-lit channel name displays above each fader, with assignable color bars. The CL1 and CL3 also feature the option of an external meter bridge.

The new Yamaha CL1. (click to enlarge)


Audio quality. Audio quality and character were top priorities in the development of the new consoles. In addition to featuring newly-designed mic preamps and delays on every input channel and output port, the range debuts a prestigious line up of studio quality processing.

The Effects Rack provides the equivalent of eight SPX2000 effects processors, along with a range of VCM EQs and dynamics, while two further virtual racks provide access to up to 32 channels of graphic EQs.

However, probably the most exciting sonic innovation is the introduction of the Premium Rack concept. Developed by Yamaha’s Dr K (Toshi Kunimoto) and his team, the Premium Rack provides a range of extremely high quality, dynamic processors and EQs.

Yamaha has collaborated closely with Rupert Neve to incorporate the acclaimed Portico 5033 EQ and 5043 compressor as key elements in this new concept. These processors are included as standard in the CL series, eliminating the need for any plug-in management.

“For the first time we have the capability of bringing Rupert Neve sound into the live audio field, entirely due to Yamaha VCM technology. I believe that it is indistinguishable from the original analog sound,” says Neve.

Scalable solution. A key factor in making the CL Series so flexible is the pair of accompanying I/O racks, the Rio3224-R and Rio1608-D, and the fact that consoles are the first to feature built-in Dante networking as a standard feature.

A scalable system is easily constructed by simultaneously attaching up to eight I/O racks via Dante, providing up to 256 input sources. Pairs of CL consoles can also be cascaded to handle larger mixing requirements.

Rio rack-mountable I/O units. (click to enlarge)


Connection of basic systems is easy, using the console’s auto-configuration facility. Two or more consoles can share the inputs from one set of I/O racks without fear of unexpected level changes due to the inclusion of Auto Gain Compensation within the I/O racks themselves.

With the new Dante 32-bit mode of operation, gain compensation can be provided without audibly affecting the dynamic range.

“We are extremely excited to be collaborating with Yamaha on the extraordinary new CL Series and I/O racks,” notes Audinate CEO Lee Ellison. “Dante provides a flexible, low latency, highly scalable, plug and play networking solution to connect Yamaha networked systems, Dante Virtual Soundcards or any other Dante networked device.

“We believe the combination of technologies integrated in this new platform will provide an unsurpassed digital experience.”

The Rio3224 also includes four stereo AES-EBU outputs, keeping signals in the digital domain right through to the amplifier. The new consoles offer three MY card slots on the rear panel, maintaining compatibility with every existing audio format as well as newer cards like the MY8-Lake speaker processing card and the MY16-Dugan auto mixing card.

The three card slots also allow for additional I/O alongside the consoles’ onboard eight mic inputs and eight line outputs.

The new CL5. (click to enlarge)

Software control. CL Editor is a new standalone online/offline editor which runs on both Macs and Windows PCs.

Featuring all the functionality that users will be familiar with from other Yamaha Editor software, it does not require Yamaha Studio Manager as a host.

Further new software applications include a new version of StageMix for iPad, which offers comprehensive wireless remote control and has been expanded to include new features such as channel naming, DCA fader control and tap tempo.

Crucially, both CL Editor and StageMix can be run simultaneously, allowing very flexible options for engineers, sound designers and system technicians alike.

Meanwhile, Yamaha’s new File Converter software has been upgraded to allow straightforward exchange of console files between the CL Series, PM5D, M7CL and LS9.

Recording. Every CL console customer will receive a copy of the new Steinberg Nuendo Live recording software, which has been designed specifically for live recording applications.

Further aspects include StageMix for iPad, the MY16-Dugan auto mixing card, and Nuendo Live. (click to enlarge)


Available from July, it includes unique features not found in any other live recording software and will be tightly integrated with CL consoles to provide optimum ease of use.

When combined with Audinate’s Dante Virtual Soundcard (also included with every CL console) engineers can easily use the Dante network to record up to 64 tracks of audio to either a Mac or Windows PC.

“Nuendo Live is not only ultra stable and easy to use, but it integrates seamlessly with the latest generation of Yamaha live consoles,” says Steinberg managing director Andreas Stelling. “This is another successful example of the strong ongoing relationship between Yamaha and Steinberg.”

In addition, basic stereo recordings can be done via a convenient 2-track USB recording and playback function.

CL5 console plus Rio I/O. (click to enlarge)


With the advent of the CL Series, the company is confident these products strike the right balance between innovative functionality and sound investment potential.

“The CL Series ideally answers today’s needs using today’s technology” says Kazunori Kobayashi, general manager of the Yamaha Pro Audio Division. “The collaboration between Yamaha, Rupert Neve Designs, Audinate and Steinberg has made it possible to deliver a sublime balance of sound, performance, and features that results in uncompromised overall mixing capability and quality. After a quarter century of evolution, the CL Series represents a momentous new chapter in the history of Yamaha digital mixing.”


The CL5 and Rio3224-D are scheduled to be released in the spring of 2012, with the CL1, CL3 and Rio1608-D available in the summer.

Yamaha

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Posted by Keith Clark on 03/21 at 06:44 AM
AVLive SoundRecordingChurch SoundNewsPollProductConsolesDigitalInterconnectNetworkingProcessorSoftwareSound ReinforcementPermalink

Tuesday, March 20, 2012

Soundcraft Extends Digital Console Networking Backbone With AVB, Dante & BLU Link Options

Get more of the latest news from the 2012 PL+S show.

 
Soundcraft is extending its range of digital audio transport option cards for its Vi, Si Compact and Si1/2/3 Series digital consoles with all options expected to start shipping this year.

Harman is one of the pioneers of AVB IEEE 802.1 and an active AVnu Alliance member, and with the standards now ratified, work has commenced to make AVB audio networking available on the Vi, Si Compact and Si1/2/3 Series consoles.

Plans are also in place for Dante network cards compatible with Vi, Si Compact and Si1/2/3 series consoles, following an agreement reached between Soundcraft and Audinate.

“This agreement is a reflection of Harman’s responsiveness to its customers who want Dante networking,” adds Lee Ellison, Audinate CEO.

To further enhance integration between Harman’s other Professional audio brands, option cards are also in development to provide a simple on and off ramp between Vi, Si Compact and Si1/2/3 Series consoles and the BSS Soundweb London digital audio bus system BLU link.

The BSS digital audio bus is integrated with the majority of BSS Soundweb London series products and with the PIP-BLU is available for a number of Crown amplifiers.

Existing expansion cards already available for Soundcraft digital consoles or stage boxes include Cirrus Logic, CobraNet, MADI, Aviom A-Net, AES, Digigram Ethersound, Riedel Communications RockNet, Alesis ADAT, SD/HD SDI and Dolby E.

Soundcraft product manager Richard Ayres states, “Connectivity is an important part of everyone in pro audio’s future; new products must integrate with legacy systems and infrastructure whilst existing products must adapt to emerging technologies. With these announcements we are making it clear that by investing in a Soundcraft console you are assured it will remain future-proof and core to an integrated system.”

Audinate
Harman
Soundcraft

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Posted by Keith Clark on 03/20 at 12:07 PM
AVLive SoundChurch SoundNewsPollProductAVConsolesDigitalInterconnectNetworkingPermalink
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