Article

Thursday, April 05, 2012

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.

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.

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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Wednesday, April 04, 2012

In The Studio: Four Tips For Mixing An Unruly Lead Vocal

If you smack the vocal around enough, it’ll behave

 
As I was telling one of the members of Mix With Us, I learn something new every time I mix a song… and I imagine that will always be the case.

Mixing is such a wild task. It’s exciting, tedious, invigorating, and depressing… all at the same time.

But there’s nothing like those final stages of a mix, when everything is finally coming together.

One of the biggest hurdles you’ll run into is the lead vocal. Your goal is to have an awesome lead vocal that sits just right in the mix. However, this proves to be more difficult than it seems.

Sometimes the vocal track has too much low end. Sometimes it’s too harsh or thin. Sometimes it disappears in the mix. Other times it’s overpowering the mix.

Be patient. Give it time. If you smack the vocal around enough, it’ll behave.

As I’ve been wrestling with my own lead vocal tracks, I’ve come up with a couple tips I’d like to share with you.

1. Use A Low Shelf

I talk a lot about getting rid of excess low end by using a high-pass filter. Sometimes, though, this can be too drastic of an effect, especially on something so prominent in the mix like a lead vocal.

If you’re having a hard time finding a good balance between too boomy and too thin, try using a low shelf instead of a HPF.

Turn the low shelf down by 6 dB or so, then roll it up from 100 Hz as high as you want, until the vocal stops being boomy but still retains its fullness.

I just mixed a song today, and I think I did a 9 dB cut with a low shelf at 300 Hz on the lead vocal. Sounds extreme, I know, but it sounded right. Had I tried a HPF at 300 Hz, it would’ve sounded WAY too thin.

2. Ease Up The Compression

Compressing a vocal is a great way to tighten it up and help it sit in a mix. However, it’s really easy (especially if you’re new to mixing) to over-compress the lead vocal.

Remember, if you remove all dynamic range from an instrument, be it a lead vocal or a guitar, you lose some of its musicality. Too much compression can kill the life of a vocal.

If you’re having a hard time getting the vocal to sound right, try dialing back the compression a bit…or perhaps don’t use any compression at all.

3. Use A De-Esser

You’ll run into this scenario quite a bit. You’ve dialed in the perfect lead vocal sound. It’s EQ’d perfectly, with just the right amount of compression, then….SSSSSSssss… The vocalist sings a word like “Mississippi,” and the S’s chop your ears off.

That’s a byproduct of compression. It tends to increase the volume of the sibilance of a vocal. The solution? A de-esser.

A de-esser is simply a compressor that only compresses a specific frequency range. It’s decided to compress (i.e. turn down) the sibilant frequencies of a vocal track without affecting the tone of the rest of the track. If dialed in correctly, the de-esser will turn down those S’s in Mississippi…and your ears will thank you.

(Be careful, though. If you overdo de-essing, the vocalist will sound like he/she has a lisp.)

4. Turn The Vocal Down

Sometimes the simplest solution is the one we don’t think about. It’s really easy to make the lead vocal TOO loud in the mix.

It’s understandable, you’ve crafted this exquisite lead vocal sound. You want the world to hear it, right? Right. But turn it down. The vocal should sit in the mix, not on top of the mix.

Hope these tips help you out next time you’re mixing a lead vocal.

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

 

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Tuesday, April 03, 2012

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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Changing Times: Just What Is A/V Over IP?

An increasingly wide range of systems are being merged into the traditional IT network architecture

 
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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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 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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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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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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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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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}

Church Sound: Acoustics… It’s All About Signal-To-Noise

Sometimes things aren't just about your system and mix

This past weekend I was at a Regency Ball with my family for an afternoon of exquisite ballroom dancing (think Jane Austen, 1812). The ballroom is located in South Bend, IN, in a restored old building called the Palais Royale. With its tall ornate ceiling and beautiful wooden floor, it’s one stunning place! 

At the ball, there was live music (piano, harp, and flute) and a “caller” who walked us through the dances. He used a handheld wireless microphone feeding a system with a number of ceiling loudspeakers placed throughout the space. 

The loudspeakers seemed to be placed well enough to deliver adequate coverage, and their overall sound quality was fine. However, the majority of the time I found it difficult to understand what the caller was saying. Why did this happen?

1) The room was reverberant; the hard smooth surfaces of the floor, walls and ceiling acted as great reflectors of sound.

2) There was a lot of additional noise created by the participants.

3) I hate to admit it but my hearing isn’t what it used to be. (Ah, you can’t beat getting old!)

While I was struggling to understand what the caller was saying, my brain was pondering the real problem. I recalled something my good friend Vance Breshears (principal consultant at Acoustic Dimensions) said a little over a year ago: “acoustics can be simply defined as signal-to-noise ratio.”
 
In the ballroom, I could hear the caller (the signal), but at times the people and reflections (the noise) were interfering with the signal. What could I do to reverse this ratio?

First, and I might add, simply, I began joining sets of dancers that were close to the caller. I figured if I was able to get close enough to hear the acoustic sound of his voice, the signal would be stronger than noise. 

Second, I worked to get myself positioned with “more mature” dancers, figuring they were probably having the same problem and would thus listen more intently and be quieter while the caller was talking.

My plan greatly helped, although being able to hear better did not mean I danced any better… :>)

I’m telling this story to get to this point: in your space, what is the noise? Is it the door at the back of the sanctuary that creaks, becoming horribly noticeable during the middle of the service? Is it the HVAC system that sounds like a wind tunnel when it fires up? Is it something simple, like the ushers forgetting to shut the doors to the lobby/narthex, allowing chatter and other unwanted sounds (footsteps, traffic noise outside the building, and so on) to drift in?

This Sunday, take a listen to the noise that interrupts the message (signal) and see what you can do to eliminate or at least minimize it. Sometimes things aren’t just about your system and mix.

Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 25 years.

{extended}

Thursday, March 29, 2012

Dynamics, Condensers & Phantoms: Getting Into The Design Of Microphones

Experiment, evaluate and make your own live sound a little better

Why are there so many microphones? Which is best? These are among the most common questions asked by my audio students.

First, there is no single mic best suited for all tasks, and this statement is also probably the best answer as to why there are so many mics on the market.

Our ears are very sophisticated transducers capable of changing sound into electrical impulses that our brain can interpret as sound.

Sound frequencies are measured in Hertz (Hz). 1 Hz is one sound wave passing a certain point at the rate of one time per second. Our ears are capable of “transducing” sound pitches ranging from 20 Hz to 20,000 Hz (20 kHz.)

Unlike our ears, mics can only accurately “transduce” (and reproduce) a particular range of pitches within the normal “20 to 20k” range.

If a mic can reproduce frequencies without boosting or cutting volume in a certain range, it is said to be “flat” in that range. This measurement of mic accuracy is known as frequency response, and is one characteristic that is a “must know” for any engineer when choosing a mic for a particular application.

A mic’s frequency response can potentially change the timbre of a sound because of its inability to reproduce all of the frequencies present in an audio signal.

Another characteristic to consider when choosing a mic is its signal-to-noise ratio (S/N) - the amount of usable audio from the instrument or vocalist as compared to the inherent noise that the mic generates by itself. (Every piece of analog and digital audio equipment has a S/N ratio and a dynamic range, which is S/N plus headroom)

But possibly the biggest factor to consider when choosing a mic is its design type. There are two basic types of microphones, dynamic and condenser.

Although dynamic mics have long been the choice of live audio engineers, condenser mics are making their way to the stage more of late for reasons that we’ll address later. And both are worth examination.

Let’s take a look under the hood.

DYNAMIC MAGNETICS

There are two classifications of dynamic mics, “moving coil” and “ribbon.” Dynamic mics work by the principle of magnetic induction. Most of us have experimented with magnets. They are bi-polar, with one side we’ll refer to as “north” and another side we’ll call “south.”

When two magnets are held “north to north” or “south to south” they tend to repel (or push away) one another. Oriented “north to south,” magnets tend to attract one another. If magnets are held in this orientation close enough to create an attraction, but not touching, a magnetic field is created between the two.

This magnetic field contains invisible lines of “flux.” All dynamic microphones have this magnetic field. In ribbon mics, a thin corrugated strip of metal is suspended between the magnets in the magnetic field. As the sound waves strike the ribbon, it vibrates. This vibration breaks the lines of flux, which induces an electrical voltage. This voltage is conducted by the ribbon and is identical to the frequency of the vibrations of the sound waves.

In dynamic moving coil mics, there is a thin diaphragm that is attached to a coil of wire. This diaphragm/coil assembly vibrates in the magnetic field, which breaks lines of flux and induces a voltage in the coil. Again, the vibration is identical to the frequency of the sound waves.

The sensitivity of dynamic moving coil mics is determined by the size of the diaphragm, strength of the magnets and the amount of wraps of wire in the coil. 

Although ribbon mics are more sensitive than moving coil mics, they are more fragile. Ribbon mics also exhibit a bi-directional pickup (or polar) pattern - they are sensitive to sound in front of, as well as behind, the mic. Couple this pickup pattern with inherent frailty and most agree this limits uses for ribbon mics on stage.

MOVING ROBUSTLY

Conversely, moving coil mics are inherently robust. This design is an overwhelming success story for the working band or sound reinforcement crew. It is also inexpensive.

Most moving coil mics are unidirectional - they only pickup what is in front of them. This is useful because sound entering the mic from the monitor system can cause that squealing sound called feedback that is an unmistakable beacon of failure. Sound engineers work hard to avoid it, and everyone in the house knows when they don’t. 

The strength of the magnets and the number of wraps around the coil matters, as does the size of the diaphragm. Large diaphragm moving coil mics are more sensitive than their smaller diaphragm counterparts. They’re generally good mics for drums with the exception of snares, which produce a high-end rattle (from their wire “snares”) that small diaphragms (found in mics like the Shure SM57) seem to do better on.

Snare drums also produce very high sound pressure levels and can distort large diaphragm moving coil mics.

Large diaphragm moving coil mics like the AKG D112, D550 or the Shure Beta 52 make very good kick drum mics. Sennheiser also makes some nice mics for drums, such as E602 for kick as well as E604 for toms. The latter also includes a mount that grips the top rim of the drum, eliminating bulky mic booms that clutter the stage and can be knocked over. (I’ve even used these in the studio.)

If you’re running sound using onstage wedges for monitors, dynamic moving coil mics are a good choice, providing a high amount of gain before feedback while also relatively easy on the pocket book. If you’re using in-ear monitoring systems, you may opt for condenser mics for vocals and some instruments. There is quite possibly some additional expense, but the rewards can be considerable.

CONDENSERS FOR STAGE

Condenser mics have found their way into more and more live sound applications. Condensers are very sensitive and have a flat frequency response over much of the 20 Hz to 20 kHz audio range, in part due to their design.

These mics work by the principle of variable capacitance, with a fixed plate and a moveable plate. These plates function as the polarized magnetic source. Sound pressure enters the mic and causes the moveable plate to vibrate in proximity to the fixed plate. This vibration is identical to the original frequency of the sound vibration. 

As the plates move closer, then farther apart, they perform the function of an electronic component known as a capacitor. The capacitance varies and a small electronic circuit in the mic produces a current flow that mimics the sound signal. Dynamic mics’ magnets are charged at the manufacturer so that they retain up to 20 percent of the voltage applied to them permanently.

Unlike dynamics, condensers don’t have magnets that are charged permanently at the time of their manufacture. As a result, their diaphragms must be charged every time they’re used, accomplished with the use of phantom power. Supplied to the mic through the mic cable, phantom power is normally 48 volts DC for large diaphragm “air” condensers.

The variable capacitance design of condenser mics makes them more sensitive to incoming sound pressure. They’re very “hot” mics. Guitarists can relate this to active circuitry in guitar pickups, and for much the same reason.

Condenser mics also generally exhibit a very flat frequency response. One specific design of the genre is the electret condenser. This design can use a small DC battery or phantom power of considerably less voltage than the 48 volts required by air condensers. Don’t confuse this function with a battery-powered transmitter for wireless mic systems.

PHANTOM CASES

Not only does phantom power provide the necessary voltage for the plates, it also provides power to the onboard impedance transformer. Without phantom power, the large-diaphragm air-condenser mics will simply not work. So where does it come from? Normally, the mixing console supplies phantom power.

Many consoles offer a separate button or switch on each channel that will enable phantom power. Some consoles have a switch that enable several channels at once, or one switch that enables all of the channels.

Fewer buttons saves cost but sacrifices function. And certain consoles don’t offer phantom power capability or only 18-volt phantom power for electret condenser mics only, making external power supplies necessary for 48-volt air condensers.

With a crisp clean sound, it’s tempting to use condensers on vocals. You’ll find numerous models designed for just that purpose. But be aware of some precautions. Condenser mics are more fragile than dynamic moving coil mics, so it’s a good idea to have solid road case for them when used on tour.

Also note that because condensers are so sensitive, they tend to be more susceptible to feedback. Care must be taken to manage stage volume. In-ear monitors can be useful in this scenario. Also be sure to make your vocalists are aware of these facts as well so that they can take some precautions. 

What if you phantom power a dynamic mic? It doesn’t damage the mic, but it’s unnecessary. Here’s a tip - if you’re using a condenser mic and notice that you’re not phantom powering it (because there’s no sound!), mute the channel before engaging the phantom power. Doing this while a channel is live can result in a loud pop.

There can never be too many mics to choose from, and inventive engineers and microphone manufacturers are discovering new techniques on a regular basis. Experiment, evaluate and make your own live sound a little better.

Scott Foulkrod has a degree in audio engineering and currently teaches audio engineering at the college level.

{extended}

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.

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.

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.

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]

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.

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.

{extended}

Capturing The Energy Of Live Shows

Audience mic techniques to enhance recordings.

What makes a live recording sound live? The audience, of course.

A live recording is all about the energy of the event, and that energy comes from the crowd, so some real thought has to be given as to how it’s captured.

Just setting up some microphones haphazardly usually produces less-than-desired results.

To avoid that scenario, let’s have a look at some proven mic techniques for live recording.

First, it can be tempting to use approaches that engineers recording classical music deploy, such as spaced pairs, X/Y, ORTF and Blumlien.

What they’re trying to do is capture the ambience of the environment and a “perfect” stereo image, but our primary concern is capturing the audience.

Note that these are two different beasts and have to be handled that way.

Sure, capturing some of the ambience is essential to a great sounding live recording, but it will come as a byproduct of a well-mic’ed audience, so it’s not important to worry about it until the primary mission is accomplished.

Audience mic’ing is a situation for omnidirectional mics if you have any, but never underestimate the value of a couple of short-scale shotgun mics.

These are especially useful because they help to attenuate the intimate conversations from the crowd that happen around where the mic is placed.

In you don’t have the option of either an omni or short shotgun, make sure that the mics that you do utilize are identical models. Also, don’t forget to engage the low-frequency rolloff switch if the mic has one.

Simple Methods
More often than we would like, we need to record some audience tracks but don’t have anything special in the way of mics or any time to experiment.

Here are some quick ways to get you in the ballpark in smaller venues like clubs. Keep in mind that the larger the venue, the more care and mic coverage is required.

Simple method 1: Place a pair of identical mics at about the halfway point between the edge of the stage and the back wall of the venue. Make sure that the mics are placed at least 3 feet above the audience.

The higher the mics reside over the audience, the better, but if it’s a club with a low ceiling, it’s better to seek placement closer to the audience to avoid ceiling reflections that can impact quality.

Start with the mics facing directly at one another across the audience as in Figure 1, then aim them both down towards - but not exactly at - the middle of the audience.

Simple method 2: For mono tracks, splay the mics off access, as shown in Figure 2. This configuration produces fuller sound in mono, but will result in a stereo track that’s off balance because one mic is pointed more toward the stage and house system than the other.

Simple method 3: At about the middle of the venue, fly a couple of mics from the ceiling, pointing directly down and hanging by their connectors, as shown in Figure 3. Many clubs use this for more permanent audience mic placement, but it works temporarily as well, assuming flying the mics isn’t too complicated.

Be sure to hang each mic the same distance from the stage as the other to keep the stereo image balanced. This approach is where omnidirectional mics can come in handy.

Advanced Methods
Let’s up our game a bit, and note that the following techniques can be used either by themselves or in combinations for best audience coverage.

Front hall: Directly in front of the stage, place two identical mics between the house system loudspeakers, pointed at the middle of the room, as shown in Figure 4 .

The trick is to find the null point in the loudspeakers, where in general you can hear them the least, and low frequencies in particular are at their weakest.

As a variation, use two mics pointed towards the center, and two more mics pointed towards the side.

This technique works great when you just can’t find anywhere secure to place mics in the crowd. And, if there’s a balcony, aim the mics at the farthest seat instead of the middle of the room.

Backline: Placing mics on the backline of the stage provides a great drum sound (not that you’re really looking for one), but more importantly for live recording, it also results in a great audience sound.

Place these on tall stands on the back of the stage pointing at the back of the singers(s) heads, as shown in Figure 5.

Rear hall: In cases where you’re already using the front hall or backline configurations, a pair of rear hall mics is usually needed as well.

Place these mics forward looking at the stage and 6 feet or so from the rear wall and/or corner of the venue so they don’t pick up any unwanted reflections, as shown in Figure 6.

At The Console
Sometimes the easiest place to put audience mics is at the front of house console, especially in a large venue.

Assuming that you’re set up in the middle of the audience and not under a balcony or some other obstruction, locate four mics (preferably shotguns) at the corners of the mix position, as shown in Figure 7.

The front two mics should aim toward the front of the house. just in front of the PA stacks, while the rear mics should point at deep house left and right – in other words, towards the corners of the venue.

This approach works great by itself, and even better with the addition of front hall or backline mics.

Further, any combination of any of these methods that cover the front, rear and middle of the audience present not only a lot of options for great audience coverage, but for capturing some really nice ambience as well (Figure 8).

Keep in mind that when using multiple pairs of mics, it’s best to record them on multiple stereo tracks to keep your variations open during mixing.

The Great Outdoors
Mic’ing a crowd outdoors poses a different set of circumstances in comparison to the indoor experience. For one thing, placement is usually a lot more difficult, with fewer options for hanging mics.

In addition, the ambience of the venue is lessened, so you usually need to resort to using more mics as a result. And don’t forget the windscreens, because nothing makes a track unusable like wind blasting across the mic capsules.

Next time I’ll discuss the pros and cons of using a DAW for live recording.

Bobby Owsinski is a veteran audio professional and the author of several books about live and recorded sound. {extended}

Wednesday, March 14, 2012

Achieving Ambience With Personal Monitors - Can You Hear Me Now?

The very isolation that makes personal monitors effective can cause problems for performers

In-ear monitors (IEMs) are radically different from the traditional floor wedge method of providing a monitor mix to performers.

For many artists and engineers, they are the ideal monitor solution. But for others, they’re simply a method of trading one set of problems (extreme volume, tiny sweet spot, feedback potential) for another (sense of isolation, can’t hear amps or audience).

Clever engineers and earphone manufacturers have come up with various methods of handling these issues, all of which revolve around how ambient sound is dealt with.

The goal of most professional earphones is isolation. Think of it in terms of signal-to-noise ratio. The intended signal (monitor mix) is pumped directly into the ear, while the undesirable noise (ambient sound) is shut out. The greater the isolation, the higher the S/N ratio.

Isolating earphones can typically achieve about 18-24 dB of isolation (with some custom designs achieving up to 37 dB). Without the competition from ambient noise on stage, the user can hear the monitor mix with exceptional clarity at significantly lower volumes. This is the source of the hearing conservation claims typically mentioned as a selling point for in-ear systems.

But it’s up to the musician to take advantage of this new listening environment by reducing volume settings.

A Good Thing?

The very isolation that makes personal monitors effective can cause problems for performers. By eliminating ambient sound, IEM systems performers can no longer hear the acoustic output of instruments (drums, piano) and amplifiers, and the hopefully rapturous sound of the audience. And on-stage communication becomes impossible without removing an earpiece.

In fact, many artists have taken to performing with one earpiece in and the other out. In terms of hearing protection, this is the worst of all possible worlds. The in-ear mix is now competing with information from the open ear, requiring at least 6 dB of extra level to be as intelligible as it is with both earpieces inserted.

At the same time, the open ear is fully exposed to uncontrolled stage levels. But if onein, one-out is unhealthy (and it is!), what’s an artist to do? There are several ways to approach this issue, ranging from clever and low-tech to sophisticated and high-tech. Let’s take a look.

Mics On The Crowd

To get crowd noise into the mix, many monitor engineers simply mic the audience and add it into the inear mix. Microphones can be set up on lighting trusses, out at the house mix position, or wherever.

Typically, these are turned down during songs and brought up in between, allowing the band to “connect” with the audience. This solution answers the issue of isolation from the fans, but has a couple disadvantages:

1) It puts an extra burden on the monitor engineer, who now has to “mix” the crowd microphones throughout the show, and eats up precious inputs at the monitor desk.

2) It’s impossible to mic an entire audience in a way that mimics what musicians would hear naturally. A stereo pair of microphones only retains natural direction cues if the artist is facing the right way, and even then the distance cues are wrong. In addition, using microphones on a crowd does not help to allow conversation on stage.

The Dave Matthews Band addressed that issue when they went to personal monitors. Monitor engineer Ian Kuhn put lavalier microphones on the performers to enable onstage communication between numbers.

Adding extra wireless channels just for that purpose is very gear-intensive, and certainly not economical. However, it does allow conversation within the band, something that Kuhn notes is critical to the band’s stage show.

Ambient Earphones

Some manufacturers have attempted to solve the ambience issue by offering an ambient earphone option. There are two types: passive and active, generally available in custom fit versions only.

Passive ambient earphones are essentially the same models sold for full isolation, but altered by drilling a “port” into the plastic shell to let a limited amount of ambient sound to enter the ears acoustically. Inside the port is a fixed 12 dB filter to limit the amount of ambience coming through.

Of course, the louder the stage, the more ambience leaks in, competing with the monitor mix, which must then be adjusted (louder) to suit conditions. On very loud stages, such level could prove harmful, so a plug is provided to close the port in situations where the ambience is too loud, or not desired.

While the passive ambient porting approach is effective in terms of allowing artists to hear stage ambience, it essentially eliminates the primary benefit of isolating earphones.

By adding an ambience port, the noise floor of the listening environment is raised (and thus, the S/N ratio is reduced) by the same amount. This means that the in-ear mix needs to be louder in order to be heard as effectively as without ambience.

As a result, artists using passive ambient earphones can hear the crowd, the stage, and each other, but without the control needed for different stage situations, and without the benefits of true isolation.

Active ambient earphones take a more technical approach. Tiny custom microphones are embedded within the earphones themselves, positioned to create a true binaural listening field. The output from these mics is added to the IEM mix in a bodypack. This is a tricky business, requiring special microphones and circuitry to avoid distortion and provide the same natural sound quality one would hear without earphones.

From a performance perspective, this is the best of both worlds. Full isolation is retained, allowing the artist to “dial in” as much or as little ambience as needed. A simple switch on the bodypack allows the performer to choose between “full ambient” and “perform” modes.

Typically used between songs, full ambient mode is, essentially, like listening without earphones, so performers can speak naturally among themselves, hear the crowd reaction, etc. In “perform” mode, ambience is reduced (or eliminated, if desired), so the artist gets the precise combination of monitor mix and stage ambience he prefers.

There’s no need to remove earpieces between songs, and no reason to engage in the (literally) deafening practice of one ear in, one ear out.

Early adopters of active ambient technology included music directors, who need to communicate with band members on the fly, and guitarists, who often need to hear their amplifiers acoustically as they utilize effects and feedback.

The goal is to provide artists with all the benefit of an isolating system, plus the ability to communicate on stage and hear those adoring fans. Which approach to ambient listening is “right” for your situation? Obviously, it depends on the resources at hand.

Using microphones on a crowd is a limited approach, but works with the tools at hand. Adding additional communication microphones to the artists is an improvement, but very complicated to implement.

Passive ambience is an imperfect solution, but may be right for some artists. The clear gold standard in terms of problem solving is an active ambient IEM system. The price may be high, but those who are using them say they are worth every penny.

Jack Kontney is a long-time pro audio guy and president of Kontney Communications, a content creation and marketing consultancy with a client base that includes Sensaphonics Hearing Conservation.

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Tuesday, March 13, 2012

Church Sound: Mastering Signal Flow – Here it Comes And There it Goes

You can have a great grasp of how your system works when you know how audio signals are routed

 
Growing up in a one hundred year-old home, I learned how to fix plumbing problems. 

One lesson was tracing all the pipes in the basement so I could tell which pipes had fresh water and which ones carried sewage. 

You might say it was my first exposure to signal flow.

Signal flow is the flow of the audio signal from the sources of input to the places of output, from sound-to-loudspeaker, if you will. 

Learning how to set up the stage, you see how the signal flows from an electric guitar to a pedal board to a DI box and then into a stage jack.  Now you will see what happens to that signal once it gets to the sound booth.

Asking For Directions

An audio signal travels from the source towards some sort of output. For example, a singer’s voice is picked up by the microphone which, through a series of components, makes its way out to the house speakers. Consider this as the general directionality of the audio signal – from a source to a destination.

The sound booth is like a giant airport where signals are coming in and going out, from multiple sources to multiple destinations. Destinations can be house loudspeakers, floor monitors, recording software, and even church nursery loudspeakers. The primary component that takes care of all these transfers is the mixer.

Coming into the mixer, are a variety of sound sources which will all be assigned to the channels on the mixer.  Naturally, you have your sources from the stage but you also have sources such as a computer, tape deck, CD player, and even audio feeds from video devices.

Once the signal is going to a channel, it tends to follow a general path that can vary slightly from one mixer to the next. From the signal input, it usually follows as such;

1) Gain control (controls how much of the signal you are letting into the system)
2) Insert loop (plugs in back of mixer for auxiliary effects)
3) High-pass filter (used to cut out frequencies below a fixed point)
4) Equalizer
5) Channel on/off or mute switch
6) Fader
7) Pan control (for stereo panning)
8) Out to groups (for control over multiple channels from one group channel)
9) Out to main fader control

Looking at the signal flow on the mixer channel, most extra on-board controls like compression and padding occur before the signal goes to the equalizer. 

Also, signal to the auxiliary controls for the channel’s audio sends, such as for monitors, will be either before or after the fader depending on it the auxiliary control is set to send as pre-fade or post-fade.

Out Of The Mixer

The signal can travel out of the mixer in a variety of ways;

—Auxiliary sends.  This could be for monitors, hallways loudspeakers, or however your system is set up.

—Tape deck. Yes, some mixers have separate send controls for tape decks. Really, they could be used for any device but yes, they might be marked on your mixer as “Tape controls.”

—Insert loops. These are at the channel level and are used to send the signal out to a processing unit like a reverb unit, and then return that sound back into the channel at the same point.

—Group ins/outs. You can send the specific group signal to a separate out.  For example, a group could be used for additional signal processing like a compressor and therefore you can route the signal out to the unit and back in. Or, route it out for some other use.

—House loudspeakers. You gotta fill the room with sound!

The output signal can go directly to another processing device, like a reverb or compressor as I mentioned above. Or, it could be routed for your house loudspeakers. It’s here where you need to dive behind your components and start following some cables.

Audio signals can be routed a variety of ways. I’ve seen systems where the main out was first routed through a tape deck before going to the amplifier. Such routing is understandable considering the fact that you commonly want to record a service. 

However, for more finite control, they could have routed the recording device to an auxiliary channel where the mix for the recorded media could have been different.

Note that the signal strength between components is considered to be at line level.

You should expect the mixer’s main output to go out to the house EQ, through any house compressor or limiter, and then to the mixer.

What To Do When The Volume Stops

Knowing how sound travels through the audio system, you can quickly find the source of audio problems. Take, for instance, an electric guitarist who has started playing during the sound check but you don’t hear them in the house mix.  By tracking the audio signal from its source, you can investigate where the problem might be located. 

For example, if you are getting a green light on the channel, you know you are getting a signal from them. Perhaps the gain is too low. Maybe you aren’t getting any signal light. Start at their guitar and make sure the signal is taking the right path to the sound booth. It could be a mistake as simple as improper cabling into a direct input box.

When The Sound Is Bad

Every time an audio signal goes from one component to another, you can pick up unwanted noise. This is called line noise. Regarding the signal flow coming from the stage, we need to look at something called the signal-to-noise ratio.  In short, this ratio explains the quality of your sound. 

For example, let’s say you have an acoustic guitar with an on-board amplifier. This gives the musician the ability to control the level of the signal from their guitar. If they don’t raise the on-board amp’s volume high enough, you could hear a large amount of noise in their signal. By increasing the volume on their guitar amp, they are improving / increasing the signal-to-noise ratio. This means when the signal is amplified, there is very little noise heard because the strongest signal is coming from the guitar.

Summary

The flow of the audio signals through your system can take different paths. There is an order to the mass of cables running to and fro. You can have a great grasp of how your system works when you know how your audio signals are routed.

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. He can even tell you the signs the sound guy is having a mental breakdown.

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