Article

Tuesday, May 08, 2012

Allan & Heath Flys The UK Flag At SXSW Festival

Allen & Heath’s iLive digital mixing system was selected by promoters, Cato Music, to manage FOH and monitors on the British Music Embassy stage at this year’s iconic SXSW festival in Austin TX, USA.

Showcasing the UK’s brightest new bands and artists, the British Music Embassy was exclusively based at downtown Austin’s Latitude 30 club, where an iDR-48 MixRack and iLive-T112 Control Surface was installed at FOH, digitally split using a Dante networking card to a duplicateiDR-48/T112 for monitors.

There were performances from nearly 60 bands over the course of the week-long festival, including emerging talent Django Django, fiN, Maverick Sabre, Benjamin Francis Leftwich, Jonquil, Ben Howard, Slow Club, Twin Atlantic, D/R/U/G/S, Skindred, and Clock Opera.

“I ran the desk in a very analog way, all inputs on the left banks, and master section on the right and some visiting engineers wanted things moved around but we always managed to get the bands ready in the 15 minute changeovers,” explains FOH house engineer, Fabrizio Piazzini. “Some band engineers had never used the desk before but were so stunned by the ease of use, sound quality and format of the system they promptly added iLive to their riders.”

“The festival patch included 37 channels but running iLive Editor meant I could drop all the channels for each performance on a single layer with just a few clicks,” Piazzini continues. “On the monitor side, we were running 5 mixes, with the odd band turning up with full IEM rigs. To ring out the wedges, we used the iLive MixPad app so we could actually EQ the system in the spot the artist would stand.”

The iLive system was supplied by Allen & Heath’s USA distributor, American Music & Sound, which also supplied Turbosound loudspeakers and beyerdynamic mics.

“I think iLive was a critical piece of the set up and it delivered every night,” commented Glen Rowe from Cato Music. “The British Music Embassy was a huge success this year and some sessions were so busy we set up more delays and opened the windows on the venue for people to hear outside!”

Allen & Heath

 

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SISME Named New Italian Distributor For Powersoft

Powersoft has announced SISME as the new distributor of Powersoft products in Italy starting May 2. Luca Giorgi, Pro Audio BU manager of Powersoft, made the announcement.

The trading company SISME has been importing and distributing musical instruments, sound equipment and HI/FI in the Italian market since 1967.

“Despite the substantial changes the modern retail business incurred in the last years, our company continues to evolve thanks to our competence and reliability,” says Claudio Bugari, Chief Executive of SISME. “Our dynamic vision of the market convinced us to make our presence heard in the segment of professional audio products. Tha t’s the reason we decided to complete the range of our amplification product portfolio with a quality brand such as Powersoft.”

The search for constant innovation, customer-centricity and high quality products are the strengths the two companies share. The new agreement signed aims at increasing the presence of green soul Powersoft designed amps in the Italian market.

With this in mind, SISME will continue to help customers find the best solution to their professional audio needs by offering its expertise in the field and a full line of products that will suite their needs, including Powersoft amplifiers. SISME distributes a full range of professional audio products including L-Acoustics, Shure, and HK Audio.

“We think we have found the right partner for the Italian market. SISME has a great distribution network well rooted in the Italian territory,” concludes Luca Giorgi. “Their payoff summarizes the spirit of our partnership: a quality company that distributes quality.”

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Peavey IPR Series Power Amplifiers Now Shipping

Peavey has announced that the IPR 1600 DSP and IPR 3000 DSP power amplifiers are now shipping. Loaded with a proprietary digital signal processing suite, IPR DSP Series amps are live production tools for both front-of-house and monitors.

The IPR 1600 DSP and IPR 3000 DSP power amplifiers combine loudspeaker management with the light weight power and performance of the original Peavey IPR power amplifiers.

The onboard digital signal processing system includes preset banks for popular loudspeaker types and configurations, as well as Waves MaxxBass psycho-acoustic processing.

The IPR DSP takes users through an intuitive setup wizard to set their EQ curves, delay speakers, crossover (full-range or full-range with sub), high pass and low pass; adjust a four-band parametric EQ and horn EQ; and specify the amount of Waves MaxxBass processing from 0-100 percent.

MaxxBass uses psycho-acoustics to calculate precise harmonics that are related to the fundamental tones of sound. When these harmonics are combined, it creates the effect of deeper low frequencies.

IPR DSP users can choose from a bank of popular Peavey and generic loudspeaker types, including models from the Peavey QW, SP, EU, Impulse, PR and SSE Sanctuary Series, as well as standard 10-, 12- and 15-inch loudspeakers. Program EQ curves include rock, pop, jazz, hip hop, contemporary worship, speech and acoustic. A built-in security lock is selectable for all functions or all but volume.

Designed with an advanced switch-mode power supply and a high-speed class D topology, the Peavey IPR 3000 DSP power amplifier provides 1,490 watts RMS per channel @ 2 ohms (840 watts RMS x2 @ 4 ohms) with a weight of just 7.8 lbs. The IPR 1600 DSP is rated at 900 watts RMS per channel @ 2 ohms (5115 watts RMS x2 @ 4 ohms) at 7.25 lbs.

Peavey IPR Series power amplifiers feature two channels and a variable-speed fan housed in a lightweight aluminum chassis. Inputs are combination ¼” and XLR, while outputs are combination ¼” and twist-lock connectors. Peavey’s exclusive DDT™ speaker protection with multi-point clip sampling leads a protection-circuitry suite that also includes DC, Temp, Signal and Active safeguards, all referenced on the front panel with LED indicators. Peavey IPR 3000 and IPR 1600 power amplifiers are backed by Peavey’s free five-year extended warranty.

The original Peavey IPR 1600 and IPR 3000 power amplifiers both won the Best New Power Amp Award from music-products retailers in the annual Music & Sound Awards, presented by the Music & Sound Retailer.

The Peavey IPR 3000 DSP ($799.99 MSRP/$599.99 street) and IPR 1600 DSP ($699.99 MSRP/$449.99 street) power amplifiers are available now from authorized Peavey dealers.

Peavey

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Five Finger Death Punch Now Touring With Midas PRO2C

As the heavy metal band Five Finger Death Punch tours North America in 2012, front of house engineer Bruce Reiter will be mixing them on his new desk: the Midas PRO2C digital mixing system.

“I was looking for a compact console and saw it online,” Reiter states. “I was considering a lot of different desks, but then I had the opportunity to touch the PRO2C for the first time at the NAMM Show. I decided then and there that I had to have it.”

Two things combined to convince Reiter to invest in Midas digital.

“First and foremost, it actually sounded good,” he says. “The preamps and EQ define the basic sound of any console, and in my opinion, MIDAS digital sounds as good as my old analog console of choice, the Midas XL3. The other thing was the way it’s laid out. It’s comfortable to mix on, and incredibly fast to get around. Everything is right at your fingertips.”

That navigation is a product of the PRO Series architecture that eliminates layers in favor of multiple channel grouping options called VCA, MCA, and POPulation groups, to enable flexible configuration and instant recall of associated channels. The PRO2C and its sister console, the PRO2, both utilize the MIDAS DL251 fixed I/O stage box, which provides 48 inputs and 16 outputs, with eight more inputs and eight effects returns available on the mixing surface.

On-board processing includes six multi-channel FX engines and up to 28 Klark Teknik 31-band graphic equalizers. The only significant difference between the consoles is that the PRO2 includes eight more physical faders on the mixing surface. “Originally, I was looking at the PRO2,” notes Reiter. “But once I tried them both, I just fell in love with the 2C. It has all the same I/O and effects, and is a perfect fit for my style of mixing.”

In fact, the PRO2C is the most compact Midas digital console available, measuring less than 35 inches wide and weighing just over 80 pounds. Working with a band like Five Finger Death Punch, that was another big selling point for Bruce Reiter. “On this tour, we have one truck and two 15-foot trailers being pulled behind two buses,” he reports. “That has to hold everything: sound, lights, backline and merchandise. Every bit of space counts, and having a console you can literally lift up and stack on top of the pack is really convenient. Being able to do that while improving the sound of the show is like icing on the cake.”

Having now had the PRO2C on tour, Bruce Reiter is more convinced than ever that he made the right choice. “I’ve mixed on just about every digital system out there and, at some point in our travels, I’m sure I will again,” he says. “But as long as I have control of it, the PRO2C or something else from the PRO Series will be my preference. Sonically, nothing else comes close.”

While most tours opt to rent their consoles, Reiter sees things a little differently. “Like most independent sound engineers, I’ve always had a certain supply of equipment that I bring to the party,” he notes. “Back in the day, it was a big rack full of effects. Later, that became a USB stick with plug-ins and licenses. This is the next step. The PRO2C takes up less space than my old effects racks, and it’s got everything I need to mix a killer show. It’s the perfect tool for me.”

Midas

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Monday, May 07, 2012

DiGiCo SD9 Adds Perfect Control To Trinity Theatre Sound

Stage Electrics has chosen a DiGiCo SD9 console to control an all-new audio system, designed for both live performance and digital cinema, at the Trinity Theatre in Tunbridge Wells, UK. 

Built in the mid 19th century, the Holy Trinity was Tunbridge Wells’ first parish church. After its final religious service in 1972 its Grade 1 listed status ensured safety from demolition, and by 1975 a public petition had secured permission from the Church Commissioners to produce a plan for community or public use.

An appeal committee raised £50,000 and five years later it reopened as The Trinity Theatre arts centre complete with a raked-seating auditorium: growing popularity soon saw an art gallery, licensed bar and computerized box office added.

Its latest upgrade sees the venerable space take on the very modern mantle of digital cinema, although a cursory gaze at the vaulted balconies, plush stage tabs and comfy seats reveals little. Only on closer inspection does it turn out to be the UK’s first digital cinema to employ the unique K-Array system, its mid/high hangs barely visible against the tabs, complemented by minuscule surround sound satellite loudspeakers discretely located around the auditorium. Supplied and installed by Stage Electrics, the system, powered by bespoke K-Array Class D high power density amplifiers with integral DSP, is controlled directly from a DiGiCo SD9 console.

“This was my first project after I joined Stage Electrics,” observes Business Development Manager for audio James Gosney. “Stage Electrics is doing bigger and bigger sound installations including the installation and supply of equipment to the Royal Shakespeare Theatre last year. As a consequence of being asked to design and supply high end audio systems, they have been expanding their audio team with people experienced in sound system design and installation, which is precisely my background.”

Coming from a family also deeply immersed in theatre, he says: “I immediately fell in love with the building. For the last 20 years I’ve been mostly involved in designing systems for big churches, so for me it was a perfect combination of the two: a theatre in a church… with a bar; it doesn’t get much better.”

“The brief was for a multipurpose theatre system, one that would work for all the types of the events that go on here,” explains Gosney. “Like jazz evenings, musical theatre, straight plays, opera, local amateur dramatic groups, pretty much everything – and on top of that, 7.1 digital cinema, with its specific Dolby processing requirements.”

The Stage Electrics commissioning team set up the DiGiCo’s system alignment and output processing with presets for cinema, musical theatre, straight plays, jazz and other types of events.

“We had shown our demo SD9 to [Trinity Theatre head technician] Simon Diaper who loved it, partly because it’s so easy to use and so logically set out, but particularly because of the sound quality, which is noiseless really.  It’s beautiful,” concludes Gosney. “And that’s the system: a DiGiCo going into the K-Array amps into the K-Array speakers, and it’s that simple. I’m all for keeping sound systems as simple as possible. Keep the signal path as clean as you can and don’t complicate it with too much nonsense in between.”

DiGiCo

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Friday, May 04, 2012

NSCA Presenting Digital Marketing Boot Camp In Las Vegas

The National Systems Contractors Association (NSCA) is hosting a two-day boot camp featuring keynote speaker, Scott Klososky, from the 2012 NSCA Business & Leadership Conference (BLC).

The boot camp will be held June 11-12 at the Las Vegas Convention Center, Las Vegas, NV.

Coming off of the most popular BLC, of which many topics centered around social media and marketing techniques, this boot camp will provide strategy-based curriculum on social and digital marketing strategies for marketing professionals and senior executives.

Attendees will understand the full breadth of resources available in the social media world in additional to the dynamics, processes and trends available today. Klososky notes that attendees will leave with “more knowledge to drive a noticeable, measurable impact on their organization’s bottom line and online marketing strategies over the next few years.”

The two-days will be spent on instruction and concept delivery in these key areas:

• Why social technologies are exploding, their role in today’s organizations and why leaders should pay attention;
• 15 unique social dynamics web 2.0 has delivered and you can apply to your organization’s strategy;
• Processes of social media implementation; and
• Future path of social technologies.

Klososky, a former CEO of three successful startup companies and current founder and Chairman of the Board of Alkami Technology, specializes in looking over the horizon with how technology is changing the world. His vision and ability to see trends in emerging technologies allow him to be a thought leader who applies his skills to help organizations thrive, leaders prosper, and entire industries move forward.

NSCA members receive this two-day boot camp at 40 percent off the original costs. Additionally, NSCA members can apply up to $400 in NSCA Education Credits towards the cost of the boot camp. 

The training includes meals, a USB drive with process implementation documents, tools and templates supporting the strategies discussed, and all the presentations for a total cost of $1,800 per participant.

For more information visit www.nsca.org/bootcamp or contact Bonnie Taylor, NSCA Events Specialist via email at .(JavaScript must be enabled to view this email address). 

NSCA

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Thursday, May 03, 2012

Low-Voltage Audio Products: Power & Noise

Meeting the challenges associated with the use of low-voltage audio information appliances.

 
This is an installment in a multi-part series. Additional segments are available here.

Noise
Low noise and low voltage don’t like each other.

Low voltage usually means portable, and portable always means low current to prolong battery life. You can design low noise and low voltage if you can be a current pig, but if you must have low noise, low voltage and low current—well, that’s difficult.

Everything works against you. The easiest way to make a really low noise op amp is to run as much current as possible through the front-end differential-pair until the silicon glows.

As unintuitive as it may be, a plain resistor, hooked up to nothing, generates noise and the larger the value the greater the noise. It is called thermal noise or Johnson noise (John Bertrand Johnson first observed thermal noise while at Bell Labs in 1927, publishing his findings as “Thermal agitation of electricity in conductors,” Phys. Rev., vol. 32, pp. 97-109, 1928), and results from the motion of electron charge of the atoms making up the resistor.

All that moving about is called thermal agitation (caused by heat—the hotter the resistor, the noisier).

Therefore quiet designs should use small resistor values, but, alas, small resistor values draw large current, and there goes the battery life. Compromise must ensue.

It is difficult to find the perfect balance between small resistor values for low noise and large resistor values for low current consumption. To make it even harder, with most analog circuits small resistor values mean correspondingly large capacitor values.

Large capacitor values do not hurt the noise performance but they are physically large and cost more, so you must make a compromise between noise, space and cost (analog design is like that).

The choice of resistor values then becomes the deciding factor in selecting the right op amp for each application. Look at the resistor values; if they are very small (like in a mic preamp) then the noise contributed by the op amp becomes critical.

However, if the application is active filters, say, and the resistors surrounding the op amp are at least 10 k ohm, then the dominate noise factor becomes their thermal noise, not the op amp’s noise. Understanding this simple fact allows you to use low-cost op amps for most of your needs.

Ultimately the performance gets down to how much voltage is available and how low is the noise floor: power supply and noise—the big two in designing quality audio for IAs.

Power Supply Design
Successful IA audio circuits begin with power supply design. Designing low-voltage audio circuits for portable and wireless information appliance products puts severe restrictions on quality.

Sacrifices necessary to keep cost, size, and weight to a minimum often hurt audio quality.

Portable and wireless devices force audio designers to work with very small supply voltages, often just a single 1.5-volt cell. There is just one rule when designing quality audio circuits if you only have 1.5 volts to work with: make more voltage.

Separate Audio Supply
No matter what the voltage, in order to achieve very high performance levels, audio circuitry must run from dedicated supplies.

Obviously it does no good to select the lowest noise op amps if they are connected to a digitally corrupted power supply.

Single-Supply Design
If the design cannot justify split-supply costs then you must design with a single supply. Since audio is an AC (alternating current) signal, its voltage swings positive and negative about some reference point.

This reference point is normally ground (or common) for a bipolar or dual power supply, i.e., one with positive and negative voltages (e.g. ±15 VDC). If you only have a single supply then you must create a reference point equal to one-half of the available supply.

For example if you have a single 5 volt supply then you create a common reference point at 2.5 volts, which allows the audio to swing ±2.5 volts (from the reference point up 2.5 volts to the +5 volt limit and down 2.5 volts to zero.

Splitting a single supply voltage is not difficult, nor expensive (although in some designs every extra op amp or resistor can mean trouble).

Techniques exist ranging from a simple two-resistor voltage divider to more elaborate buffered op amp designs. Excellent application notes covering all aspects of this topic are available from Texas Instruments, Linear Technology, and Analog Devices.

DC-DC Converters
If the hand you’ve been dealt contains only one AA cell battery then you must become a DC-DC converter designer at once. Luckily there is lots of help in this area. There’s nothing you can do with a single AA battery except use it to create more voltage.

How much voltage depends on the product and the application. If you must create loud audio into big speakers, then life’s going to be a lot harder than if you can get away with driving only headphones.

Low efficiency loudspeakers and headphones are a big obstacle to pristine IA audio. Low efficiency means you need lots of power to drive high-quality speakers to loud levels. And lots of power means lots of voltage and current.

If it is your choice, then chose a pair of nice clean and quiet split supply voltages—as high as you can get them for loud results or if you are going to interconnect with the pro audio world. Most pro audio products use ±15 VDC for their analog audio circuits.

While finding a single IC capable of converting 1.5 VDC to a nice clean and quiet ±15 VDC is difficult (see LTC Design Note) to impossible, several IC companies make converters that will pump up 1.5 volts to 12 volts, and from there you can split that into a useable ±6 VDC. See for instance Analog Devices or Linear Technology, or also Linear Technology.

See also Linear Tech’s latest free design software for DC-DC converters, although it doesn’t help much for single cell converters.

Another free helpful DC-DC converter design program is available from National Semiconductor named Switchers Made Simple , and take a look at the collaborative venture by National, Vishay, and Pioneer-Standard Electronics called Webench , a free on-line tool to design, simulate and order prototype kits for power supplies.

And not-for-free from ON Semiconductor is Power 4-5-6 software for the design, simulation and analysis of power topologies.

Op Amp Specifications Important For Audio
Selecting op amps for audio is a lot easier than it was the first time I wrote about this topic in 1976 (Audio Handbook, National Semiconductor Corporation, 1976. The reprinted version is the last revision published by National Semiconductor in 1980, compiled and edited by Martin Giles who took over as compiler and editor after I left in 1976. Order copies from Old Colony Sound Lab) .

This is primarily due to the quantity of audio specific chips sold into the automotive and PC industries.

Quantity is what IC companies understand. They live and die by quantity, and for the first two decades, audio was pretty much ignored as a product line. Back then selecting good audio op amps took some digging and required the designer to know quite a bit about audio’s specific requirements.

Things are different now. Audio-grade op amps are sold by the millions each day, and it makes selecting them a lot easier since most IC companies have a separate section in the selection guides for audio.

Here is a summary of the most important parameters (in no particular order):

• Gain-Bandwidth Product, or GBW, equal to at least 3 MHz. This gives plenty of open loop gain (>40 dB) for feedback circuits to still work well at 20 kHz. More is better as long as the phase margin does not get compromised. You want to see a solid phase margin of 60 degrees at the unity gain BW crossing point.

• Slew Rate, or SR, equal to at least 1.5 V/microsecond. This value is necessary to prevent slew-limiting at 20 kHz with full output voltage. In a single-cell world you never have large voltage swings so you never need large slew rates, but it’s nice to have some margin.

• Noise, or Noise Density: normally specified at 1 kHz, along with a graph showing wideband performance. Look for spot noise density at 1 kHz less than 15 nV per square-root-Hz (approximately the noise of a 10-kohm resistor) for low gain circuits (like filters) and less than 4 nV per square-root-Hz (noise of a 1-kohm resistor) for high gain circuits (like mic preamps).

In addition to a low 1 kHz spot noise number, you want to see a low 1/f corner, i.e., you don’t want the low-frequency noise to start rising dramatically until below 20 Hz.

• Total Harmonic Distortion + Noise, or THD+N: This is not a spec to get overly concerned with. As long as the part stays out of whole numbers, you probably don’t have to worry about any audible results. But in the interest of successful marketing, select parts with a THD+N less than 0.1% over the entire 20 Hz - 20 kHz audio range. Today it is very hard to find parts that don’t shine in the THD department.

Low noise, high slew rates, wide bandwidths, and excellent linearity (low distortion) characterize high quality audio op amps. Other important specifications are application driven and include power supply voltage, current consumption, common-mode rejection, power supply rejection, input impedance and size.

The Audio Handbook (see above) describes op amp audio requirements as follows: “The IC must process complex AC signals comprised of frequencies ranging from 20 Hz to 20 kHz, whose amplitudes vary from a few hundred microvolts to several volts, with a transient nature characterized by steep, compound wave fronts separated by unknown periods of absolute silence.

This must be done without adding distortion of any sort, either harmonic, amplitude, or phase; and it must be done noiselessly—in the sun, and in the snow—forever.” Nothing has changed.
Selecting Low-Voltage Op Amps

Good audio requires good parts. Low-voltage information appliances make selecting the right audio ICs even more important—and more difficult. What follows are guidelines and pointers to high-quality audio ICs specifically designed for low voltage designs.

Note: There are too many world wide semiconductor companies to be all-inclusive regarding recommendations. Apologies are made to those left out. The author knows the ICs and companies spotlighted from direct experience. Omission of any company or specific products merely means the author was not aware of them. It is also recognized that many of the ICs mentioned will be outdated immediately upon writing, so always check the manufacturer for the latest part replacing or improving the one discussed.

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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Transcending Tech: A Conversation With Ethan Winer, Author Of “The Audio Expert”

Getting down to how it really works

The Audio Expert, a new book by Ethan Winer, exhaustively covers a plethora of important technical aspects of audio. But it goes much further, discussing and explaining the relationship between audio and a wide range of closely related factors. In short, it challenges you to think, to seek a deeper understanding.

Just released by Focal Press (and available here), I received an advance copy and have had a hard time putting it down. Winer, who has worked with audio for more than 40 years, is a mix engineer musician, product designer, author (and more), and in 2009, he presented the Audio Myths seminar at the AES Convention in New York that’s still generating buzz.

I recently caught up with him to discuss the book as well as a range of other topics.

KC: What was your primary motivation for writing the book?

EW: Two reasons – one is to dispel the many myths I see repeated endlessly in audio magazines and web forums. Most aspects of audio science have been understood fully for more than 50 years. Yet some people still believe that competent wires can sound different, that typical amounts of phase shift are audible, that jitter is a problem, that digital “summing” in DAW software is somehow flawed, and so forth.

Almost daily I see posts in audio forums by people with limited funds asking if they really need to spend a lot for a microphone, preamp, converter, or external summing box to get professional results. So my goal is as much consumerism as education, to help people spend wisely.

The other reason is to explain how audio really works to those who are interested. Forty years ago, recording engineers were as much “real” engineers as they were recordists. Back then, most knew how to solder up a patch bay and align a tape recorder, and many could read schematics and do at least minor repairs. George Massenburg is a perfect example – he’s renowned for the quality of his recordings, as well as for designing the first parametric equalizer.

When I started recording professionally in the 1970s, audio magazines included technical articles and DIY plans, and manufacturers were proud of their high fidelity and provided specs for distortion and frequency response. Today, a loudspeaker review is likely to state the size of the woofer but not its frequency response, which of course is what really matters! And you almost never see distortion specs or off-axis response. If an active loudspeaker includes distortion specs, it’s usually for the power amplifiers only, not the complete system.

Many mix engineers have the talent to make music sound great, but without understanding the engineering and science behind the gear they use. I appreciate that some people don’t care at that level, but many do. In my estimation, the pro audio press has let us down in this regard, dumbing down content, and even perpetuating many of the same myths you read in hi-fi type magazines.

Your experience is more with recording than live sound. What’s the value of the information you’re providing for the live sound practitioner?

The Audio Expert is a comprehensive “reference” type book covering all aspects of audio, so there’s plenty for everyone – even interested audiophiles. It’s written for people who want to understand audio at the deepest, most technical level, but is presented using plain-English explanations and mechanical analogies with minimal math.

Besides describing how many different audio devices are used, it also explains how they work internally. The book brings together the concepts of audio science, aural perception, musical instrument physics, acoustics, and basic electronics, showing how they’re intimately related. So while I don’t address directly the challenges facing live sound engineers, there’s a huge amount of educational content. It’s definitely not a “Dummies” type book for beginners!

If you could recommend one chapter as the “must read” of the book, what is it, and why?

Perhaps most important is explaining in great detail how fidelity is defined, with included audio examples people can play on their own systems to determine at what level distortion and other artifacts are audibly damaging. This is addressed mainly in Chapters 2 and 3, though this type of information is sprinkled liberally throughout the book.

Besides the 65 demo audio files available on the book’s web site, there are also 31 videos and five audio-related software programs.

What’s the single biggest misconception or “myth” about audio?

The two biggest myths are probably that there are aspects of audio fidelity that “science” hasn’t yet learned how to measure, and that listening is a more reliable way to assess the quality of gear than measuring. I see magical thinking all the time in audio forums, but it’s easy to prove that everything affecting the fidelity of audio devices is already known.

A spectrum analyzer can display artifacts 100 dB below the music, and is highly reliable and repeatable, versus human hearing that varies from moment to moment, and is influenced by the masking effect. Many types of distortion and other artifacts can be very difficult to hear, even when they’re only 40 dB below the music.

What sources proved most valuable as you wrote and assembled 650-plus pages of significant technical information provided in the book? How did you fact check and verify?

The book actually totals 739 pages when including the three bonus chapters online. I’ve been involved with audio for many years as a recording engineer, circuit designer, and computer programmer, so I already had a solid grasp of the science. But I did learn a few things! I was fortunate to get advice from microphone expert Bruce Bartlett and loudspeaker expert Floyd Toole.

Another friend, electronics engineer John Roberts, read my entire manuscript as I wrote it, and audio expert Mike Rivers did the technical review. All of these people provided invaluable suggestions and fact checking.

How do you clearly separate what is objective in audio versus what is subjective?

Subjective preference is impossible to define, so I don’t even try. I do address some aspects of preference, such as the perceived improvement after adding acoustic treatment. But mostly I address the science of audio, and explain how audio circuits and their plug-in equivalents are used and how they work internally.

It’s impossible to “measure” the quality of a piece of music, or assess one’s enjoyment. But it’s absolutely possible to assess fidelity, even when a perfectly clean sound is not the artistic goal.

In your view, what are the differences between analog and digital audio in terms of sound quality?

First we have to define what is meant by analog and digital. “Analog” encompasses both audio hardware such as equalizers and compressors, as well as the recording mediums of magnetic tape and vinyl records.

Digital audio refers to both the recording medium and software effects. Analog gear can be very high quality, with distortion and noise low enough to not hear, and a frequency response flat enough to not matter.

Gear that meets these criteria is considered audibly transparent, such that it’s difficult to notice a change in quality after passing through the device.

Digital plug-ins have a slight advantage because their transparency is dictated entirely by the resolution of the math used to perform the needed calculations. Most modern software processes audio data using 32-bit floating point numbers. This is potentially cleaner than any real-world electronic circuit.

Of course, every computer sound card and outboard A/D/A converter has analog input and output sections, and these ultimately limit the fidelity possible. But many converters are audibly transparent. So the real answer is that both analog and digital can have acceptably high fidelity when implemented properly.

Another important factor falls outside the context of “sound quality” – intentional subtle distortion used for effect to add faux clarity to a track or complete mix, or as “glue” to make a mix sound more cohesive. A compressor with both the attack and release times set very fast also adds distortion that is useful in some contexts. These effects can be implemented effectively using either digital or analog technology.

What are the best ways to learn the essential principles of audio?

The best way to learn is by doing. It also helps to have knowledgeable friends, whether in person or a web forum. Of course, the downside of web forums is having to sort through many disparate opinions to separate fact from belief. But anyone who has basic audio software can easily try things for himself or herself.

Often I’ll see someone in an audio forum ask, for example, if they should compress before EQ or vice versa, or use EQ boost rather than cut – even though it would be trivial to just try it to find out for yourself firsthand!

If a tree falls in the forest and no one is there to hear it, does it make a sound?

Yes!

For a much more detailed and interesting answer to that last question, be sure to check out The Audio Expert, published by Focal Press (ISBN: 9780240821009) and available here.

And, go here to read an excerpt chapter entitled Audio Fidelity, Measurements, And Myths - Part 1, provided exclusively to PSW.

Keith Clark is editor in chief of ProSoundWeb and Live Sound International.

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Lectrosonics Introduces ASPEN Dante Network Processor

Lectrosonics has expanded the ASPEN digital matrix processor family with the new SPNDNT network processor, a full-featured DSP that can address both the ASPEN and Dante matrices and add mixing, gain, and delay functions to the digital audio signals.

Any of the 48 final mixes in the ASPEN matrix can be assigned to any one or more of the Dante channels for transport to other endpoints in the network.

Dante signals can also be imported into the ASPEN matrix in a local processor sub-system for mixing with other local and network signals, and then routed onward to other local or network devices.

Dante products deliver a no-hassle, self-configuring, true plug-and-play digital audio network that uses standard Internet Protocols.

This combination offers a market leading solution today, while providing a migration path to upgrade to new standards such as the IEEE Audio Video Bridging (AVB).

Gordon Moore, vice president of sales at Lectrosonics, states, “Dante has proven to be robust, reliable and incredibly easy to configure. Our customers have long asked for a low latency, easy to use, audio transport over Ethernet system.

“Dante delivers all that is promised,” he continues. “To say that we are excited about the potential is an understatement.”

The ASPEN Dante network processor offers a solution in system designs for telepresence, room combining, courtroom complex, distance learning and multi-endpoint conference systems. Available now, it carries a MSRP of $3,325.

Lectrosonics

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Tuesday, May 01, 2012

New Mackie DL1608 Digital Live Mixer In The Field With Dierks Bentley

The Mackie DL1608 digital live mixer with iPad control, introduced at Winter NAMM this past January, is being field tested in anticipation of its release this summer, including the launch of the Master Fader control app in June.

Recently, the DL1608 got some face-time with mix engineers for Dierks Bentley and the band Brad (Stone Gossard, Shawn Smith, Regan Hagar, Jeremy Toback) at a live show at Seattle independent music store Easy Street Records.

The live show was part of national Record Store Day, an annual celebration where 700 independently owned record stores collaborate with musicians to celebrate the art of music.

Easy Street Records offers a small venue with a maximum capacity of 750 people that commonly hosts afternoon shows with touring artists scheduled for headlining shows at larger Seattle venues later that evening. 

Front of house engineer James “Pugs” Mcdermott (Dierks Bentley) and Barrett Jones (Brad) had time for only a quick tutorial before their 30-minute sound checks. “Although I had never used this mixer before, it was straight forward, easy to use, and well laid out,” states Jones.

Besides having an intuitive interface that was easy to navigate with little instruction, another benefit of the DL1608 is the ability to control the mixer from anywhere in the venue. “It was great to be able to tweak the monitors from each monitor position on stage with the iPad controller. The ability to mix the show while walking around the floor was really nice,” Jones notes.

During the 10-hour day, which featured performances by four bands in total, the DL1608 performed without a hitch and Mackie received hands-on insight from the engineers. These types real world field tests are an essential part of Mackie’s product design process. Stress testing the hardware in a variety of conditions ensures the DL1608 hardware will work reliably for customers. In addition, real world feedback on the Master Fader control application will shape future updates as it continually evolves through simple app store updates.

The Master Fader app will be available free in the iOS-App Store in June 2012. Mackie will also soon premiere a DL1608 video podcast series offering tons of great information about the DL1608, including all the basics, advanced how-to’s, and tips and tricks. Later episodes will specifically respond to questions that customers pose to Mackie’s Facebook and Twitter accounts.

The DL1608 video podcast series will start in May 2012 in iTunes. The DL1608 will begin shipping in summer 2012.

Mackie

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Midas Consoles Resounding Success At Echo Awards

A Midas PRO2 live audio system at FOH and PRO6 on monitors, provided by rental and production company SoundHead Company, commissioned by production companies EBS-Lights and Showplan and sourced from German MIDAS distributor Mega Audio, took centre stage at this year’s Echo Awards, Germany’s premier music awards show. The Berlin event’s star-studded line-up included performances by Katy Perry, Lana Del Rey, Sean Paul, Taio Cruz, Rammstein and Marilyn Manson.

Soundhead Company MD and FOH engineer Tim Ehrenfried, who only recently added the PRO2 to his inventory, explains, “In the past I have had excellent experiences working with MIDAS XL8 and PRO6 consoles, and I knew I wanted that great warm MIDAS sound for the event in recognition of the calibre of international acts performing.
But the PRO2 at FOH gave me the added advantage of all the features of the big consoles in a practical compact footprint.”

Ehrenfried set up some 40 channels with wireless microphones for each artist, with additional inputs coming from all of the live acts and 10 playback tracks from the OB truck. He created individual scenes for each act and used rehearsal time to adjust and save all settings and patches. The PRO2’s user-friendly design was one of a number of stand-out features for Ehrenfried. “Speed is a big issue in our business,” he says. “You don’t often have much time to create your setup, but working with POP(ulation) groups makes the whole thing very clear and the number of I/O boxes enables you to configure your system with both hardware and software individually

for each application. Everything is where you expect it to be, and with just a few steps I could create a big set up with many channels.”

Having created and saved his settings in advance, Ehrenfried was free to enjoy mixing on the night, taking full advantage of the PRO2’s design, features and flexibility. “Best of all I liked being able to view all busses at once and to have easy access to all channels,” he says. “The dynamic EQ, which I applied to the vocals, is a wonderful device, as is the internal compressor with its many available modes.
It sounded great and it is the real MIDAS sound - very warm, and the EQ could be analogue.”

The Midas sound may have inspired the choice of PRO consoles, but it was reliability that confirmed their deployment for the prestigious event, which was also broadcast and streamed live to an international audience.  “Next to the sound it is very important that I can trust the system to deliver,” says Ehrenfried, “and that is just what these consoles do.”

Midas

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TC Furlong Inc. To Host Yamaha CL Chicagoland Launch Tour

Yamaha Commercial Audio Systems, Inc. and TC Furlong Inc. will host a mini-demo tour featuring the new Yamaha CL Digital Console Series, May 1-2, at three separate Chicago-area sites. Yamaha Systems Application Engineer, Kevin Kimmel, Regional District Manager, Mike Eiseman, and TC Furlong staff will be on hand to demonstrate the new console. All events are free of charge and open to audio professionals are welcome.

Dates and location of events are: Tuesday, May 1—2:00 PM-8:00 PM, TC Furlong Inc., Lake Forest, IL; Wednesday, May 2—9:00 AM-1:00 PM, American Management Association, Rosemont, IL; and Wednesday, May 2—3:00 PM-8:00 PM, Tribeca Flashpoint Academy, in Chicago.

The Yamaha CL console is Dante network-based and uses Yamaha CentraLogic technology. Features include remote I/O for a faster, more responsive Yamaha system solution. The three CL models (CL1, 3, and 5) are only differentiated by frame size and input capability.  All three models feature 24 mix buses, 8 matrix buses, stereo and mono buses, and 8 DCAs. The console series was specifically designed for sound reinforcement applications, including performing arts venues, theaters, houses of worship, touring, and remote broadcast.

More information
Yamaha Commercial Audio Systems, Inc.

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The Black Keys Maneuver First Headlining Arena Tour With DiGiCo

The Black Keys, aka Dan Auerbach and Patrick Carney, have had a big underground following for over a decade but with the success of 2010’s Grammy awarded Brothers and last year’s wildly successful El Camino, they’re in the midst of their first headlining arena tour across the globe.

The tour is also the first time the Black Keys have had the luxury of carrying their own production gear, which includes a DiGiCo SD10 for FOH and an SD8 at monitors, spec’d by longtime engineer Jason Tarulli and provided by Eighth Day Sound. The console’s flexibility and fidelity have given Tarulli and monitor engineer Fabian Quiroga a much-desired consistency from gig to gig.

“I was originally intrigued the first time I had a chance to get behind a DiGiCo board at a show we did in Cleveland around 2009,” Tarulli recalled. “Unfortunately, at that time, we were not carrying audio production of any kind and I never knew what I was going to get into from day to day. But on that day back in 2009, both the PA and DiGiCo SD7 desk were supplied by Eighth Day Sound (also out of Cleveland), and the tech walked me through some of the features and flexibility of the console. I immediately wanted to see more of this desk. Once we got through the basics and I was able to actually hear my mix through the SD7—paired up with a d&b J-Series rig—I was very, very pleased with how it sounded. It was a welcome moment of clarity in a mess of flavor-of-the-day consoles and PA’s.”

It wasn’t until 2011 that the Black Keys started getting big enough to carry full audio production. When asked what he wanted to carry, Tarulli immediately thought back to that show in Cleveland and began asking colleagues what they thought about the DiGiCo desks.

“I spoke with a few other engineers, including Kevin Madigan, whom I’d met through our tour manager at a festival while he was FOH for the Smashing Pumpkins, as well as Jay Rigby, the monitor engineer for Cage the Elephant. They both said essentially the same thing: ‘Get the DiGiCo desk!’ I got in touch with Owen Orzack at Eighth Day and he invited me up to their HQ and I was able to get some hands-on time with the new SD10.

“There were a few things I wanted to sort out before taking out a desk that I had only used once or twice in the past: I wanted to be familiar and confident with the functionality of the board, I wanted the capability of getting a multitrack recording of each show as well as playback for reference through the console, and I wanted to be able to check out the new Waves rack—all of which I was able to do thanks to the user-friendly and flexible layout of the desk and, of course, once again, all of the helpful people at Eighth Day.”

As they have in the past, guitarist/vocalist Auerbach and drummer/vocalist Carney tour as a four-piece, supported by John Wood on keys, rhythm guitar, percussion, and vocals, and Gus Seyffert on bass, rhythm guitar, and vocals. Tarulli is managing approximately 40 inputs from the stage, including 11 mics on the drum kit, a mic and DI on the bass, and each of the six guitar amps has a mic and a direct box patched between the amp output and speaker. Additionally, there are three vocal mics, one for percussion, and two ambient mics set up stage left and stage right for the multitrack recording.

He keeps the setup simple and uncomplicated, relying on some of the console’s key features from presets to built-in plug-ins.

“I haven’t been much into using snapshots with this band. With the way these guys operate onstage and how their flow and feel of each song can differ from show to show, I prefer to follow along as we go rather than get the rug pulled out from under me when the guys decide to change things up on the fly.

“Having said that, I still absolutely rely on saving and recalling presets for all of the built-in comps and gates as well as the Waves plug-ins that I use, as well as having the ability to build macros to control things like delays and reverbs. I also love experimenting with things like bus compression and setting up different control groups for different instruments and vocals. Essentially, I am a huge fan of flexibility and being able to adjust quickly on the fly and having the ability to save and recall it all any time that I need to.

“Generally, I use the DiGiCo built-in comps on most of the channels, as well as the built-in gates for drums. I like the Waves SSL comps on snare drum and the SSL Strip inserted on any buss I may use. I also use a touch of the DiGiCo DDL on vocals from time to time. I keep a very simple approach and try not to get buried in plug-ins and FX, and use them as sparingly as possible to keep the signal path as clear as possible.”

Monitor engineer Fabian Quiroga adopts the same approach in his third tour with the band. With extensive, previous experience working with DiGiCo desks from D5s to SD7s on tours ranging from the Ringling Bros.
Circus to Columbian pop star Juanes, he’s found the SD8 familiar and easy to get around on.

“I keep it very minimal for these guys because they’re very minimal when it comes to their sound; they want to keep it as raw as possible,” he says. “I’ve been able to get most of what I need onboard. I’ve found the multiband compression is helpful and it’s my first time really experimenting with it. Macros are great for all kinds of effects and fading and it’s one of the greatest features of the board. I have 11 inputs coming from Pat, which is average for drums, but from Dan I have 10 inputs from his guitars alone.

“Some of the guitars I only use on certain songs and it’s good to be able to select what channel you want to mute. Same with Gus and John. They start the show with all four of them, but halfway through, Dan and Pat do about 5 songs solo and it’s just a matter of hitting one macro key and I can mute them or unmute with a press of a button.”

To date, only half of the four-piece is on in-ear monitors, with Auerbach and Carney still preferring the floor wedges and side fills.

“I just put Gus and John on ears this run,” Quiroga explains. “They both provide a lot of backup vocals and a lot of falsetto, and with the wedges being so close to their microphones, there was only so much that I could push them. I suggested they go on ears and it’s been a good transition. Pat and Dan are still on wedges and I’d never push ears onto them. They like to work off each other live, and because their music is so free and open, a lot of elements would change if I forced ears on them. But it’s really loud onstage; Dan’s got four guitar cabinets and they’re all pointing directly at me on stage left.

So I decided to go onto ears as well because I can’t expose myself to all that loudness all the time. I’m mixing about 50 percent of the show using them. We’ve got these new Ultimate Ears UE 18 Pros and they sound amazing.”

One of the most basic yet effective features for Quiroga is the talkback feature. “It’s the simple things that really matter and make your day-to-day routine easier, and this is very convenient.”

Having the ability and convenience to manage the consistency of the audio production in this new flush of the band’s success has been a boon for both engineers—and has had a noticeable effect on fans to band alike. And with international dates scheduled through the fall, it should be a smooth ride from here on out.

“Ultimately, I chose the console because of its flexibility and because of its fidelity,” sums Tarulli. “Initially, the challenge for me was to get consistency.  In the past, not having the luxury of carrying any audio production, the best I could do was advance a list of things I would like to have and hope for the best. The ideal situation never really seemed to be the norm… ever.

“After a few years of essentially being thrown in front of just about every possible scenario (good and bad), the DiGiCo definitely stood out as one of the best overall-sounding desks. Once we began carrying gear with us, including the DiGiCo SD10s, everything became much more consistent night after night. We’ve also gotten more compliments about the way the shows are sounding, too… more than we have before. I would like to believe that the DiGiCo has been a part of that. I love it, and I know the band loves it too.”

DiGiCo

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Monday, April 30, 2012

Production Company MARIUS Produções Chooses Soundcraft Vi6 Digital Consoles

The municipality of Praia, located on the Cape Verde Island of Santiago, isn’t the most obvious place to find a pair of Soundcraft Vi6 digital desks.

But it is here in the archipelago where Mario Bettencourt has been operating his rental company, MARIUS Produções—the largest pro audio and lighting company on the islands—for the last 13 years, and Soundcraft desks have been a central feature.

The island hosts a population of around 500,000, and although 20 percent live in the Republic’s capital of Praia, events supported by MARIUS take place throughout the country. These include a variety of music festivals, concerts, live TV and radio music shows, theatre and dance, expecting ever higher production values.

Responding to this, and the faster changeover times required by the new standard of shows now appearing on the island, Mario placed the order for the two Vi6s with Portuguese-based importers, Tecnimusica, who provide the company with all the support necessary.

“As we are involved in practically all the big events that take place in Cape Verde, the need to make these fast stage changes and keep records of all the soundchecks in order to satisfy the artists’ demands has become paramount,” Bettencourt says. “We can be providing sound reinforcement for up to six bands a night in a festival situation, which is why I chose a sound desk that could enable me to fulfill this.”

“I asked some of my technician friends who had worked with different digital mixers,” he adds. “Hardiness, versatility, size, weight, design, ease of use, and quality of the processors were all carefully considered. But the Vi6 is the reference mixer in various riders that I receive.”

Since the population of Cape Verde is made up of Creoles, the recent Kriol Festival, which marked the debut of the Vi6, saw a big turnout. “By organizing an event of this magnitude we can make our Festival a real reference for all Creole people,” Bettencourt says. “With national music headed by Cesária Évora, along with the exposure we get to a lot of world music, Cape Verde is now a real cultural melting pot.”

Before establishing MARIUS, Mario Bettencourt had already worked with a number of Soundcraft platforms, including the Series 5, Delta and Spirit. “I always liked the sound and today own an MH3 and EFX 12, as well as the two new Vi6s,” he said. MARIUS also has various BSS, dbx and Lexicon processors in its inventory, as well as AKG microphones.

Soundcraft
Harman

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In The Studio: Laying Down Quality Drum Tracks Without A Drummer

Everything is perfect, except for one little thing...

OK, you all know the drill. It’s time to lay down the drum track for your future Grammy Award-winning song, but there’s just one tiny little problem; no live drummer within a 50-mile radius of your studio.

Or how about this one? You’ve managed to post bail (again) for your drummer, only to find out that he had to sell his acoustic kit to pay this month’s rent.

No problem, you think to yourself, I’ll just plug in my Drum-O-Matic 5000 and it’s a go! After all, drum machine technology has come a long way in recent years, and we all know about the improvements in sampling since the Drum-O-Matic 3000 was first released. 

So you get your little magic “drummer in a box” hooked up, program all your parts the way you always wanted your drummer to play them, and record your homage to John “Binary” Bonham. 

Everything is perfect, except for one little thing… your drum tracks sound horrid.

Before we continue, please let me share something with you. Although I spend most of my musical time locked away in the DAW dungeon, and these days I’m lucky if I get to play my drums once or twice a week, I am at heart a drummer.

I love the drums. To me, every great recording begins and ends with a great drum track, and likewise, every crappy recording begins and ends with a crappy drum track. 

It seems that rarely do songs get placed in the “classic recording” file without containing a well-executed, beautifully captured drum track. Unfortunately, many home recordists do not have Jim Keltner’s home number, nor do they have the budget, space, or tolerant neighbors for an acoustic kit in their home studios.

I should also say that in addition to my passion for all things drum shaped, I also have a fondness for drum loops and electronic based rhythm tracks. If executed properly, these types of sounds and grooves can create a certain type of atmosphere that is unique to their makeup.

On the flip side, canned samples, bad MIDI tracks and boring old drum machine patterns can also suck the life out of any song faster than anything.

So, at this point you’re probably asking yourself, “can I create great rhythm tracks when live drums are simply not an option?” Is that not-so-cheap Drum-O-Matic 5000 on my desk worthless? Will this article contain actual tips on building useable “non live” drum tracks? Why did so few people actually enjoy the movie “Joe Versus The Volcano”?

The answers are yes, no, yes and I don’t know…

Tip #1 - How to get the most out of your drum machine sounds
Let’s start with the samples themselves. Sometimes, you don’t want a snare drum to sound like a snare drum as much as you want it to sound like a garbage can filled with BBs being struck with a frozen fish.

While I can appreciate the need for such a sound, for the sake of this article, let’s assume that most of us are typically looking to recreate natural, acoustic drum kit and cymbal sounds. 

That said, often the collection of standard single note “hits” provided in your drum machine can leave something to be desired. If drum machine can import samples from another source, such as your PC, great, your possibilities are endless. 

If you didn’t have the money for the Drum-O-Matic 5000, so you opted for the less-expensive 3000 model without inputs…don’t panic, you still have options!  I’ve found many great software drum machines available for the personal computer that function just like their hardware big brothers. 

A few years ago, I came across a shareware product called LeafDrums, and still use it today. It’s very easy to operate, uses .wav as it’s native file source, and has a great price point!  Even if you’re stuck with the stock samples that came loaded in your drum machine or keyboard sequencer, you still have a good chance of finding a handful of sounds that can be very useable.

Tip #2 - How to program effective drum machine parts.
Let’s look at levels and panning. To me, nothing sounds more revolting than a drum machine track with ultra loud cymbals and toms that are panned hard right and left. This is not how an actual acoustic drum kit sounds when played live, nor is it typically recorded and mixed in this fashion. Sampled cymbals and toms are usually the weakest link in the chain, so be careful of how much attention you draw to them.

Also, cymbals usually do not have to be very loud to cut through a mix. Try and let the frequencies in which cymbals resonate “react” to your mix before boosting their volume. This will most likely take practice and much restraint. Toms, as well as cymbals, should be used sparingly, and should also take a back seat in the volume department. 

The kick and snare should almost always be the loudest drums in the mix, and should usually be panned to the center. As I mentioned before, beware the urge to “over pan” the toms and cymbals.

Most drum tracks (electronic or acoustic) sound best when mixed in mono, or slightly panned, with a stereo reverb applied.  Now that your samples are selected, and their levels and pans are set up in a very musical way, let’s look at how to build a great sounding drum part.

When all else fails, use the K.I.S.S. method - Keep It Simple Stupid. Try to avoid programming “busy” and overly complex drum parts.  Less is usually more, and you’ll most likely find that simple drum parts sound more natural, and will sit much better in your mix.

Tip #3 - Drum loops… Drum loops… Drum loops… Drum loops…
Often sonically superior to “perfect” sounding drum machine parts, drum loops can have massive amounts of groove, simply because they’re usually created by REAL drummers.  Whether they’ve been created from sampling famous drum tracks, or found on one of the hundreds of royalty free drum CDs and electronic files now available, drum loops may just be the answer you’re looking for. (By the way, Johnny Rockstar, loops aren’t just for hip-hop anymore.)

In my opinion, loops should have a place in every modern studio.  With the rise of DAWs and loop editing software, recordists are finding new and different uses for loops of all shapes and sizes. Loops will usually take some tweaking, but can be a very satisfying replacement, or accompaniment to a great number of drum tracks.

Tip #4 - The percussive touch
I’m willing to bet my Johnny Mathis Christmas albums that many of you are dissatisfied with the electronically manufactured drum tracks you’ve previously heard or created.  I’m also certain that at some point you’ve cried out in frustration “that drum machine part sounds too stale and mechanical”!

Live drums have a very “human” feel to them, and the slight inflections and imperfections in time, swing and stroke velocity are what give them that feel.  Drum machines and loops are limited in how they can lessen their “perfect” delivery.

Some drum machines and sequencers have modes that allow the user to randomize tempo and velocity according to taste, and this can be very useful in the fight to eliminate the stale perfection of manufactured drum tracks.

Another very useful method is to record live hand percussion, such as shakers, tambourines congas and bongos, over the top of your machine and loop parts. Obviously, this may not work in all forms of music, as you rarely hear a good maraca performance in a death metal song, but percussion played by an actual human being can certainly help in many cases.

Tip #5 - It’s all in the mix
How you mix electronically created drums within the track is one of, if not the most important aspect of weather or not you find success in this area. With regard to electronic drum tracks, the two most common mistakes I hear most often are, drum tracks that are too loud in the mix, and tracks that are drenched in reverb. 

Somewhere along the line, someone started a nasty rumor that if a particular instrument sounds lousy, you should immediately apply a large plate reverb on it. Rest assured my friends, sometimes all that will get you is the same lousy sound - but in the Grand Canyon. 

Reverb is great, loud drums are great, but be very careful not to “over dress” your drum tracks in an attempt to make them sound more “real”. 

Also, never underestimate the power of EQ and compression. Try to think of your drum track as one instrument that contains many balanced components, rather than a ton of segmented sounds.

Since overhead and room mics play such an important role in recording acoustic drums, one way to help achieve a live kit sound is to set up a room mic in your mixing room and blast the soloed drum track through your monitors. 

Then you simply mix your newly created “room” sound with your existing drum tracks.  I’ve also found that running my electronic drum parts through a distortion box or SansAmp can yield some very interesting results. The bottom line is don’t be afraid to experiment!

Tip #6 - The Internet is your friend
The web is a great resource, not only for finding tons of samples and loops, but also for finding drummers you can “cyber track” with.  I’ve collaborated with many musicians that I’ve never even met face to face thanks to my DAW and my cable modem. Not to say that you need either of these things to do host a little on-line recording session. 

Simply find a drummer on-line (throw a rock on this or any music related site and you’ll hit a dozen of ‘em), get him/her to play along with a rough mix of your song.  When they’ve completed their drum parts, they simply send you an electronic copy of the track(s) via email, snail mail (with the tracks on a CD), or via a file-sharing site. Then, Import the drum tracks into your song and presto!

Nothing can replace a good ol’ fashioned live drummer playing a well-tuned, high-quality kit.  But those of you that simply don’t have the resources to employ, feed and care for such a beast have some reasonable alternatives today.

Now go make some music!

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