Mixer

Tuesday, March 11, 2014

In The Studio: Condenser Or Dynamic For Vocals? The Cases For Both

Article provided by Home Studio Corner.

 
Whether it’s a documentary on your favorite band, a movie scene in a recording studio, or a full-page ad in Sweetwater‘s latest catalog, one common theme exists: vocalists use large-diaphragm condenser microphones.

I’m not a big fan of the phrase “that’s how we’ve always done it.” Certainly we should learn from the experience of others, but doing something just because everyone else does it leads to a fairly boring experience.

Do I use a condenser mic on vocals? Sure…but not exclusively.

I’ve mentioned this before. On my album, I used three different microphones — a tube condenser mic, a regular condenser mic, and a dynamic mic. Why? Because I chose the mic that sounded best for that particular song. Some days one mic just didn’t sound that great. I switched it out, liked what I heard, and moved on.

More and more, I reach for the dynamic mic when I need to record a vocalist. Here are three reasons why.

Less Room Noise
If you record in a noisy room, then you’re constantly battling picking up noise in your recordings. Whether it’s the computer fan, hard drives, lawnmowers outside (or inside)—it’s a common problem.

Condenser mics are wonderful. They’re detailed and crisp, but they sometimes pick up everything.

Because dynamic mics are less sensitive than condensers, you can record the vocalist without all the extra noise. This reason alone should be enough to convince you to try it.

Less De-Essing Needed
Sibilance can make or break a lead vocal track. Condenser mics tend to really emphasize the S’s and T’s of a singer. Typically, you’ll reach for your handy de-esser plug-in…but they don’t always work perfectly.

A dynamic mic, on the other hand, doesn’t capture all that extra high frequency material, and it tends to not need a de-esser at all, even with heavy compression.

Less Likely to Be Harsh
Dynamic mics don’t have the high end of condensers. That’s a given. In some scenarios this may sound dull or dark, but I submit to you that condensers can sound overly bright and harsh at times.

If you’re getting a harsh sound with your condenser, it may be time to switch to a dynamic. It won’t have nearly the high end detail of a condenser, but it probably won’t have the harshness either.

One final point on dynamic mics, if you don’t have a decent preamp with a lot of gain, you may have problems. Dynamic mics have a much weaker signal than condensers, so you need to have a preamp that can give you enough gain without adding a lot of additional noise.

Three Reasons For Condensers
Condenser microphones are, by far, the most common type of microphones to track lead vocals.

While I do love a good dynamic on a vocal, sometimes the session calls for a condenser. I would say I use a condenser microphone on a lead vocal at least 70 percent of the time.

So you may be asking how do I decide when to use a condenser? Let me share with you three reasons.

Less Gain Needed At The Preamp
Condenser microphones require a voltage in order to work. That’s why you send 48-volt phantom power to the microphone from the preamp. This voltage charges a metal plate which, when moved by sound waves, generates an electric current. This current is the audio signal.

That’s probably more detail than you wanted to know, but the point is this: condenser microphones have a higher output than dynamic mics. As a result, you don’t need as much gain at the preamp to get the signal to a useable level.

The problem with dynamic microphones is that they need a lot of gain at the preamp. Cheaper preamps typically don’t have enough gain, or introduce too much noise when cranked all the way up. You don’t usually have this problem with condenser microphones.

Lots Of Detail
Because we’re dealing with a charged metal plate and an extremely thin diaphragm, condenser mics capture much more detail than dynamic mics. This makes a condenser mic ideal for capturing the subtle nuances in a singer’s voice.

This may not matter if you are recording a screaming rock singer, but for the singer-songwriter ballad, it’s perfect.

Defined High-End
If there’s one thing a dynamic microphone lacks, it’s high frequencies. If you look at a frequency chart for a typical dynamic microphone, you’ll see that the frequency response starts to roll off around 12-15 kHz.

If you’re wanting a nice, breathy vocal sound, you probably won’t get it with a dynamic mic. When people talk about having a vocal track with a lot of “air,” chances are the vocal was recorded through a nice condenser microphone.

 
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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Posted by Keith Clark on 03/11 at 02:59 PM
RecordingFeatureBlogStudy HallEngineerMicrophoneMixerSignalStudioPermalink

DiGiCo Announces 2014 Masters Series Tour Dates

First five locations of this year’s hands-on training sessions announced

DiGiCo has announced the first five locations for this year’s DiGiCo Masters Series Training Tour, which presents hands-on training sessions.

Presented in conjunction with Group One Limited, DiGiCo’s U.S. distributor, the single-day course is designed to provide attendees with plenty of hands-on console time, including multi-track playback to the desk and the ability to mix down to headphones, allowing them to put into practice what they’ve learned throughout the day.

“The idea behind these one-day trainings is to not only give those that attend a great introduction into the DiGiCo mixing environment but also make sure they understand the ins and outs of system configuration and operation,” says Ryan Shelton, sales and support specialist and instructor for Group One. “This is a great opportunity for those interested in what DiGiCo has to offer as a mixing platform as well as those who are familiar with DiGiCo but may simply want to brush up on what’s new.”

The first two training tour dates of 2014 will take place in Atlanta on March 19 and 20, followed over the next couple of months with stops in St. Louis (April 22, 23), Chicago (April 29, 30), Phoenix (May 13, 14) and Denver (May 20, 21). Additional cities and dates will be announced soon.

Although there is no cost to attend the DiGiCo Masters Series training sessions, reservations are limited to ensure that all attendees are able to spend sufficient time on the desks. Complimentary lunch will also be provided. For more information and to reserve a seat for one of the upcoming dates, visit DiGiCo’s Masters Series Facebook page at www.facebook.com/digicomastersseries.

DiGiCo
Group One Limited

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Posted by Keith Clark on 03/11 at 08:57 AM
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Monday, March 10, 2014

Tech Tip Of The Day: What Goes Into A Church System?

10 significant aspects to keep in mind with seeking a new sound system for a church...
Provided by Sweetwater.

 
Q: I’m on a committee to purchase a new sound system for my church. Are there any special considerations that go into this type of installation?

A: In many ways, designing an effective system for a house of worship is one of the most demanding jobs in the audio business.

While you are undoubtedly interested in good stewardship of your congregation’s funds, keep in mind that the following points are not “luxuries,” but are essentials for good sound system.

#1 Dynamic Range: Church sanctuaries are usually quieter than other gathering places. In fact, the noise floor sometimes resembles recording studio environments more than auditoriums.

So, the sound system must be quieter than usual to prevent audible noise in the audience area. You should specify a system with as much as 96dB of dynamic range.

#2 Signal-to-Noise Ratio: Many listening environments have a “sweet spot” for which the sound system performance is optimized. But in a house of worship, every seat must be optimized for adequate signal-to-noise ratio. Generally a minimum of 25dB S/N ratio is appropriate for every seat in the audience area.

#3 Uniform Coverage: Many auditoriums are plagued with “hot” and “cold” spots in sound coverage. This can usually be attributed to interaction between multiple loudspeakers, and is unavoidable when more than one loudspeaker is used to provide sound coverage. A good design assures that there is even coverage in the audience area, and that no seats are unusable because of loudspeaker interaction.

#4 Versatility: While it is possible to design sound systems that are optimized for speech or music, your system must perform well for speech and music. The attributes of these two types of systems are often at odds, so this is a very difficult task.

Your system must have the accuracy and clarity needed for speech reproduction, while maintaining the extended frequency response and power handling required for music.

#5 Hum and Buzz: Audible AC hum is a major detriment to a church sound system. It usually results from improper grounding practices, either in the wiring or the actual equipment. Remember that off-the-shelf equipment must often be modified to work without hum.

#6 Gain Before Feedback: Whenever a microphone is placed in the same room as a loudspeaker, the potential for feedback exists.

Things that aggravate this further are multiple microphones and long miking distances - necessities for most churches. Your sound system must be extremely stable, meaning that loudspeaker array design and mic placement are critical to the end result.

Your sound personnel must understand the limitations of the sound system and be trained to manage the open microphones and working distances for people using the system.

#7 Wireless Microphones and RFI: These can adversely affect the performance of a sound system. It must be properly shielded against such, with appropriate filtering devices installed when necessary.

In addition, the operating frequencies for your wireless mics must be carefully selected to work properly in the presence of other RF broadcasts in your area.

#8 “Clean” Installation: An important yet often overlooked aspect of sound system design is the installation. Proper interconnect practices must be carried out, and all applicable electrical codes must be observed.

In addition, a “clean” installation means that wiring has been concealed as much as possible, and that the finished system blends well with the decor of the building.

#9 Professional Equipment: Selecting marginal equipment is usually false economy. You need a system that provides reliable, quality performance for years to come.

It’s best to deal only with companies that provide reliable, repairable products. Loudspeakers should be “stress tested” for safety, so they can be suspended above a congregation with confidence.

#10 Calibration, Training and Documentation: A properly calibrated sound system will be much easier for your personnel to operate. A significant amount of expertise is required to make a system “user friendly.”

Your sanctuary is a critical listening environment for speech and music. Your sound system must provide adequate gain, intelligible speech, even coverage and extended bandwidth to all listener seats. The best value in a sound system is one that meets all of these criteria.

 
For more tech tips go to Sweetwater.com

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Posted by Keith Clark on 03/10 at 12:57 PM
Church SoundFeatureBlogStudy HallAnalogAVDigitalEducationInstallationInterconnectLoudspeakerMixerSound ReinforcementSystemPermalink

Soundcraft To Host “Mixing with Professionals” Sessions At Prolight+Sound 2014

Training and demonstrations on Soundcraft’s Si Expression range and the new Vi3000 digital console

Harman’s Soundcraft is hosting its “Mixing with Professionals” (MwP) training sessions at the upcoming Prolight+Sound show in Frankfurt. The sessions will offer formal training and demonstrations on Soundcraft’s Si Expression range and the new Vi3000 digital console.

“We trained around 2,000 engineers over the past year on the Expression range as well as Vi and know the Vi3000 especially will draw lots of interest at the show,” says Keith Watson, marketing director, Soundcraft Studer & AKG, Harman Professional. “The MwP sessions offer a great opportunity to get hands-on experience with industry experienced engineers on hand to help and provide tips of the trade, everyone is welcome to join a session.”

The “all-in-one-box” Soundcraft Vi3000 digital live sound console offers a host of features, a new industrial design, 96 channels to mix, and onboard Dante. It also uses the new internal Soundcraft SpiderCore DSP based on sister company Studer’s Vista 1 engine, with Soundcraft’s Vi Version 4.8 operating software offering the new “3D” Vistonics user interface, while adding a fourth 24-channel fader layer to improve access to the console’s 96 input channels.

The surface operation and layout is similar to other Vi Series consoles, providing a familiar feel while offering expanded functionality. And, the Vi3000 also includes upgraded microphone preamps and 40-bit floating-point DSP processing.

The MwP sessions will be held all four days of the show (March 12-15) in both German and English languages at the Harman stand, number F32 in Hall 8. More details here.

Soundcraft
Harman Professional

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Posted by Keith Clark on 03/10 at 12:07 PM
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Friday, March 07, 2014

Yamaha Helps Reincarnate The Water’s Edge After Super Storm Sandy

Yamaha CIS Series delivers complete sound system solution approach

Destroyed by Hurricane Sandy in 2012, the Water’s Edge, a regionally known restaurant/bar/catering facility on the Barnegat Bay in Berkeley Township, NJ, has now been restored from the ground up and outfitted with a new sound system headed by the Yamaha CIS (Commercial Installation Solutions) Series.

Calling on the services of Boulevard Pro of Ridgefield Park, NJ, the Water’s Edge co-owner Dave Ziegler requested a completely designed audio solution.

“We believe in the Yamaha CIS Series complete system solution approach, and that is how I presented the new system recommendation to our client,” states James Cioffi, co-owner, Boulevard Pro. “Our experience and reputation combined with the Yamaha support network secured our involvement on this project.” Boulevard also recommended and installed point of sale, CCTV and DTV systems.

The Water’s Edge consists of three large indoor catering spaces including the restaurant, and is also used for wedding and large events. An 8,000-square-foot outdoor space houses the Tiki Bar and Edge Live, which when weather permits, features live music. Patrons arrive all summer long by land and by sea.

“Our major challenge in designing and installing the new system was time itself,” says Ziegler. “We had to have a functional Tiki Bar for the summer or we risked losing our loyal customers. Our other major challenge was complying with the sound ordinance and containing the sound within the four wind-break glass walls that surrounded the Tiki Bar that literally sit on top of the water.

“With the help of Boulevard Pro, we were able to design several zones throughout the Tiki Bar to manage the sound levels out of respect for our neighbors. James Cioffi was one of my most patient vendors, able to do all of this with NO structure in place, plotting zones on our ever changing architectural drawings, and continually modifying the design until we had a final set of drawings.”

The indoor space is outfitted with a Yamaha CIS MTX3 audio processor, two DCP1VS keypads, and 48 Yamaha VXC4W ceiling speakers. The system controls eight audio zones and is using all 12 of the MTX3 inputs.

“THE MTX3 is super powerful and sounds great,” says Cioffi. The Yamaha keypads are located in the head end interior location of the space and in the outdoor bar/lounge area, controlled completely via iPad/iPhone.

The waterfront bar/lounge system consists of a Yamaha LS9-16 digital audio console, Yamaha TX4n amplifier, Yamaha DXR 12x2 monitors, and IF2112M95 speakers. The Yamaha StageMix App on the LS9 runs live entertainment via an iPad.

“The entire venue is beautiful and really impressive, and the Yamaha system is working great,” Cioffi adds.

“We are proud to say that we are the premier spot on the Barnegat Bay hosting the best entertainment both inside and out,” concludes Ziegler. “Bands and DJ’s couldn’t be happier coming to play at our venue because they no longer have to haul all of their heavy equipment in to play. It’s a win-win situation. When my general manager isn’t around, I like to take this Ferrari of a sound system for a true test drive.”

The Water’s Edge
Boulevard Pro
Yamaha Commercial Audio

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Posted by Keith Clark on 03/07 at 03:32 PM
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Church Sound: Understanding Mic, Line & Speaker Signal Level

This article is provided by ChurchTechArts.

 
The topic for this post came out of a very real conversation I had with my new audio volunteers. We got to talking about various signal levels that we deal with in the world of audio, and it became very clear that they had not yet been exposed to any of this nomenclature. I figured it’s entirely possible that some out there are also unclear on what all these terms may mean as well.

So this will be an introductory course on basic signal level. To keep it simple, I’m not going to go into all the background math that gets us here, or define every term; I’m going to stick with the practical implications of signal level. With those two caveats, simple and practical, let’s begin.

Level And Resistance
Two terms that need defining are level and resistance. For the purposes of this discussion, level refers to voltage. Voltage is analogous to water pressure. More pressure, more flow. Voltage is a measurement of the amount of force behind the signal traveling down the wires.

The voltage we’re dealing with is pretty small, so to make it easier for us sound guys to deal with, we don’t talk about it in terms of volts at all. Rather, we use the decibel, or dB. A dB is a unitless scale used to compare like values; in this case voltage.

We have a reference value, in this case 1 Volt = 0 dBV (the V signifies voltage), and we reference everything else to that voltage. Thus, -60 dBV corresponds to 1 millivolt. Someday I’ll go into the math on how we get there, for now, take my word for it. And I encourage you to do some research on your own to learn more about voltage.

The other term we need to know is resistance, which is measured in ohms. Resistance (often expressed as impedance in our world—and yes I know that impedance is actually DC resistance plus capacitance; I’m trying to keep this simple) is exactly what it sounds like, the resistance, or impediment to signal flow.

Going back to our water analogy, think of impedance as the size of the pipe you’re pushing water through. Obviously it’s a lot harder to push a large volume of water through a straw than it is a 4-inch-diameter pipe. In audio-land, we have two basic values of impedance, low (roughly 250-600 ohms) and high (roughly 1,000-10,000 ohms). At this time, don’t worry too much about the exact values, just get the concepts. Impedance matching is a whole ‘nother post.

Right, so we have that down? Signal level (dBV) and Impedance (low and high). With that as our backdrop, let’s consider the three most common types of signals we face in audio; mic level, line level and speaker level.

Mic Level
Think of a mic level signal as a low-level, low-impedance signal. Mic level is nominally around -60 dBV, so we’re looking at 1 mV (mV=millivolt, or 1/1000 of a volt, or .001 V), give or take. The impedance is also low, in the area of 250-600 ohms.

Now, even though the voltage is low, we can send mic level signals a reasonably long distance because the impedance is fairly low. When sent over a good balanced cable, mic levels will travel hundreds of feet and arrive pretty much intact.

We see mic level coming from mics (obviously) as well as DIs (direct injection boxes). A DI turns unbalanced, high impedance signals into a balanced, low impedance mic level signal so it can be sent a longer distance.

I’ll deal with balanced/unbalanced signals in another post; for now think of them this way—balanced = 3 wires = better for longer runs, unbalanced = 2 wires = good only for short (2-15-foot) runs.

Line Level
Most professional audio gear using line level signals runs at +4 dBV, which corresponds to a little over 1.5 V. Whereas mic level signals are almost always balanced, line level can be either balanced or unbalanced.

Line level is typically high impedance, on the order of 1,000 ohms (1K ohm) or so, but because the signal level is so much higher than mic level, we can send it long distances (at least if it’s balanced).

I want to pause here for a moment to consider some practical implications. Let’s say you plug a line level signal into an input that’s designed for mic level. What would happen?

Going back to our numbers, you have an input that’s looking for 1 mV and you shove 1V into it. That’s about 1,000 times as much signal as it’s expecting (see why we use dB instead of volts? We can say 64 dB instead of 1,000 volts). You don’t have to be an electrical engineer to guess that the result will not be pleasant. While the input it not likely to be destroyed, the audio signal will be. Gross distortion will be the audible result.

On the other hand, if we plug a mic level signal into an input that is expecting a line level signal, what might happen? Again, we’re feeding a signal that’s roughly 1,000 times lower than expected; so the result will be low signal level and high noise. Starting to make sense?

Speaker Level
The third common type of signal we deal with in audio is speaker level. Speaker level is very high level (+22-33 or more dBV and can range from 11-89 volts) and very low impedance (4-16 ohms typically).

With that kind of signal level on hand, it’s pretty clear why we don’t want to plug a speaker level signal into a mic level input. That might actually blow something up. And it’s also why we can’t drive a speaker with the output of a microphone, at least not directly.

Now we could talk about the intricacies of these signaling levels for the next two weeks, but I’m trying to keep the post length manageable. In the meantime, do some research on your own. You’ll be amazed at how much more of audio makes sense when you have a firm grasp of these concepts.

Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.

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Posted by Keith Clark on 03/07 at 03:03 PM
Church SoundFeatureBlogStudy HallConsolesInterconnectMeasurementMixerSignalSound ReinforcementPermalink

Soundcraft Unveils New Vi3000 Digital Console

Includes the internal DSP Soundcraft SpiderCore, a new industrial design, 96 channels to mix, Dante compatibility and more.

Soundcraft has announced the introduction of the new Vi3000 digital live sound console, offering numerous features, including the internal DSP Soundcraft SpiderCore as well as a new industrial design, 96 channels to mix, Dante compatibility and more.

“The new Vi3000 represents a massive leap forward in digital console design and functionality, while also keeping the best elements of the Vi4 and Vi6 consoles,” states Andy Brown, Vi Series senior product manager, Soundcraft. “We have a totally new internal core based on the highly successful Studer Vista 1 and a completely new, more efficiently designed surface running updated operating software.

“Compared to other consoles in its price range,” Brown adds, “the Vi3000 offers more input channels, more busses, more faders, and a user interface like no other. We are proud to say the Vi3000 is simply unparalleled in its category.”

The Vi3000 utilizes the new internal DSP SpiderCore based on the company’s Vista 1 engine, with Soundcraft’s Vi Version 4.8 operating software, while adding a fourth 24-channel fader layer to improve access to the console’s 96 input channels.

The surface operation and layout is similar to other Vi Series consoles, providing a familiar feel while offering expanded functionality. The Vi3000 also includes upgraded microphone preamps and 40-bit floating-point DSP processing that further enhance sound quality.

The Soundcraft Vi3000 has an all-new appearance with a more efficiently designed control surface, 36 faders, 24 mono/stereo buses and a sweeping black screen panel with four Vistonics II touchscreen interfaces with sleek, updated 3D graphics. Because the Vi3000 has four touchscreens, it can be used by two engineers at the same time.

“Visibility of parameters and easy access to them are the most important features a console can have,” Brown notes. “When something unexpected happens—as it always does in live sound—being able to see a lot of settings quickly is a massive advantage.”

The Soundcraft Vi3000 also offers extensive rear panel connectivity. In addition to a full complement of analog and digital inputs and outputs, the console provides MIDI, USB, Ethernet, DVI out, Dante/MADI record feed outputs, redundant power supply and other connections.

Two expansion bays that can be fitted with MADI Stagebox cards, to connect multiple Soundcraft Stagebox input expander modules. The Vi3000 can also accommodate the new Soundcraft Realtime Rack, a hardware/software unit designed in collaboration with plug-in manufacturer Universal Audio that provides access to 74 industry-standard UAD plug-ins. And, it’s the first Soundcraft console to incorporate a Dante interface as standard, for seamless digital audio networking with Dante-enabled devices.

The Vi3000 retains all the features that have made the Vi Series a popular choice, such as FaderGlow illuminated faders that display different colors according to function, the ability to store and recall snapshots and cues, compatibility with Soundcraft’s ViSi app that allows remote control from an iPad, built-in Lexicon reverb/delays, dbx compression, effects and more.

image

 

Soundcraft
Harman Professional

 

 

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Posted by Keith Clark on 03/07 at 11:34 AM
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Allen & Heath Releases Version 1.4 Firmware Update For GLD Digital Mixers (Includes Video)

Introduces a suite of dynamic processors and enhancements to scene management, including crossfading and embedded scene recalls

Allen & Heath has released v1.4 firmware for its GLD digital mixer range, introducing a suite of dynamic processors and enhancements to scene management, including crossfading and embedded scene recalls.

V1.4 includess two multiband compressors (3 and 4 bands) and a 4-band dynamic equaliser, recently developed for the latest update to the iLive digital series.

V1.4 also includes Transient Controller, an accurate model of the industry standard transient signal processor. These additions provide GLD users with crucial tools for dynamic control, such as taming vocals, smoothing a bass line, or shaping the transient behaviour of a snare drum.

In response to user requests and especially useful for theatre production, v1.4 also provides the ability to set a crossfade time of up to 20s per each scene, which ensures a smooth transition by automating the levels and pans when the scene is recalled. And, new to scene management is the ability to recall multiple scenes. A delay time can be set for each embedded recall, and scenes can be recalled on other mixers on the network via TCP/IP, which is useful, for example, when using a separate console for submixing.

Users of DAW software also benefit from the addition of dedicated MIDI strips, which can be assigned anywhere across the four layers for simple DAW control over Ethernet. This allows direct assignment or ‘learning’ of the strip fader, rotary and keys in the DAW. Alternatively the free DAW Control driver for Mac OS X can translate the set of default MIDI messages to popular HUI or Mackie Control protocols.

Other improvements in v1.4 include 12/18/24 dB/oct variable slope on HPF filters, refinements to the I/O patching screens, and the addition of Soft Keys to the accompanying GLD Remote and GLD OneMix iPad apps.

Finally, v1.4 has offers multilingual capabilities, as the Touchscreen user interface and Help can now be displayed in different languages. Spanish and Mandarin to be available soon as a free download.

GLD v1.4 is available to download at www.allen-heath.com.

 

 


Allen & Heath

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Posted by Keith Clark on 03/07 at 10:43 AM
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Crestview Church Gets Creative With PreSonus

AudioBox 1818VSL for small project studio and new StudioLive 24.4.2 console for the sanctuary have resulted in big changes

As worship pastor at Crestview Church in Boulder, CO, Greig Hess has long been a strong advocate in creating a contemporary service that inspires through music, with PreSonus gear making a tremendous impact on his work.

“I originally got an AudioBox 1818VSL for our small project studio about a year now,” he says. “I began using it on some live events as well, and I was really impressed with it. So we recently got a StudioLive 24.4.2 console for the sanctuary, and it’s dramatically changed the way we create and record our music.” 

Hess adds that the StudioLive has enabled them to radically streamline their live service production. “The processing power in the StudioLive is equal to several times the outboard gear we had before—we can have a compressor and effects on every channel if we need to. Plus we were able to remove a whole rack of gear from front of house. And the ability to save full snapshots of everything has done amazing things for our workflow.”

As has the ability to record the entire mix, he notes. “We use Capture to record a multitrack of our live service. With a volunteer staff, it’s great because when you are training a new sound guy you can use it to show them how to run the console, how to add effects, what different things sound like. And you’re doing it in an empty room, without the pressure of a live service.”

In fact, he adds, it’s a useful thing even for a more experienced mixer. “I’m usually on stage playing, and working with a Capture session gives me a chance to get familiar with the board, so I know what to ask for when I need something. It’s also great for rehearsals - we use it to control the mix from the stage, which enables us to schedule rehearsals without having to have a sound guy present.” 

Not surprisingly, those Capture sessions have seeded other creative projects as well. “We load the files into Studio One and do some additional production, add instruments and do some sweetening to the tracks,” he says. “You can do some amazing things in Studio One.”

The sanctuary is outfitted with Renkus-Heinz TRX81 and TRX62H two way loudspeakers, powered by Crest and Crown amplification. A Hear Back personal monitoring system drives the musicians’ in-ear systems. And Hess reports he’s also picked up an AudioBox USB to use when playing tracks and loops from his laptop on stage.

Meanwhile, he says, the AudioBox 1818VSL has continued to be a workhorse on its own. “We’ve been using the AudioBox 1818 VSL in our Fellowship hall. It’s enabled us to add a 16-input interface in there, with no fuss and no setup woes. Sometimes we do video projects, and the Audiobox 1818VSL makes a great portable setup we can take anywhere. And I’m already thinking about how we can use it when we do retreats. I can basically set it all up for them here, configure it all the way they will be using it, and they can take it with them and not have to worry about needing a sound guy.”

Hess concludes, “For a smaller church like ours, PreSonus gear has really given us an edge. It’s affordable, it’s easy to use, and it gives us access to the kind of technology that only the really big churches used to be able to afford.”

PreSonus

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Posted by Keith Clark on 03/07 at 06:07 AM
AVLive SoundRecordingChurch SoundNewsAVConsolesMixerProcessorSoftwareSound ReinforcementStudioPermalink

Thursday, March 06, 2014

Church Sound: Mixing Like A Pro, Part 6—The Channel Strip

This article is provided by CCI Solutions.

 
In previous articles in this series (here), we’ve spent considerable time on EQ and gain, but this time we’re going to pick it up a little bit and cover a number of other buttons and knobs that typically exist on each channel. These all exist on a digital console too, but may not be in the same order as we’ll tackle them here.

PAD
Most of the time you won’t likely need this, but occasionally inputs send a much stronger signal than usual and you run out of room to turn the gain down. Engaging this button will provide a cushion, usually 20 dB, which supplies room to go up or down with gain without having way too much input.

Phantom Power (48V)
Phantom power is required to operate certain types of microphones and is usually supplied by the mixing console. While we’re not looking to cover all types of microphones in this article, we’ll make a distinction between dynamic and condenser mics for the sake of our discussion on phantom power.

Dynamic microphones like the Shure SM57 and SM58 are relatively inexpensive, durable, moisture-resistant and less prone to feedback. Condenser microphones tend to produce a higher quality sound (flatter and extended frequency response) and are more sensitive to picking up sound.

Condenser microphones are good at picking up more of the detail and nuance of acoustic instruments and vocals. They also require power, and that’s where our “48V” button, otherwise known as phantom power, comes in. You might have the gain set correctly and the fader set to a normal level, but if the phantom power is not turned on, there won’t be sound from condenser mics.

AUX (or Mix)
Just as faders are used to mix the house send, the Aux sends are simply another mix that can be put together. Working the exact same way the faders do to create a mix, turn the AUX knobs to increase or decrease the level of input sources into each mix.

For most people, Aux sends will feed monitor wedges, in-ears or effects. Regardless of where the final send goes, AUX sends are simply a different way to mix inputs into an output.

PAN
If mixing a stereo house, one where both the left and right loudspeaker can be heard from most seats in the house, the pan knob can help in creating a little bit of space in the mix and create a stereo image for those listening. 

When operating with a mono system, or a stereo system where each side of the house only hears one of the loudspeakers, it’s best to leave the pan knob at center so everyone gets to hear the entire mix.

MUTE
Simple enough, this button will eliminate that channel’s audio from its output destination. On some consoles (Yamaha especially), the mute button is replaced with an “on” button. In that case, switching the “onn” button off will eliminate audio.

PFL/AFL/Solo
Primarily known as Pre-Fader Listen and After Fader Listen, this button is also known as Solo. Pressing it providesthe opportunity to monitor only that input in your headphones for checking for anomalies or other specific things you’re hearing.

Some mixers have a Solo with the ability to choose whether to hear the solo pre-fader (the input right as it comes into the mixer and after the gain knob) or post-fader (the input with channel strip processing and the channel fader volume applied).

Assign/Subgroups
The assign buttons allow you to route the signal or sound of that channel directly to the master output or to a subgroup. The more technical term for a mixers subgroup is VCA (Voltage Controlled Amplifier) and the digital mixer version is DCA (Digital Control Amplifier).

When mixing 24-48 inputs, it can be tough to keep up with the dynamics of all the live musicians when dealing with each fader individually. Creating relevant groups by assigning multiple channels to a single subgoup allows you to adjust that group of channels with just one fader.

For example, let’s say we have eight inputs for our drums, bass, acoustic, electric, two stereo keyboards, a variety of orchestra instruments, six vocalists and a choir. In order to make mixing all of those inputs more manageable, we’ll assign them to the subgroups.

One possible breakdown for grouping could be:

1) Drums
2) Guitars
3) Keyboards
4) Orchestra
5) Lead Vocals
6) Background Vocals
7) Choir
8) Playback sources (CD, i-device, DVD player, etc.)

While you may prefer a slightly different arrangement (which is fine), right off the bat mixing has been streamlined through the use of subgroups. Finding that background vocals are getting a bit lost, but the blending of them is solid? Just push up the entire group a bit.

When you hitting that big accapella section of the song with just the drums, push just the drums and vocals a bit with two faders instead of grabbing 12. If one song is guitar led and the next one is keyboard led, make that adjustment quickly as well, without changing the overall balance of what is in each group.

If you’re going to be an active sound person (and I hope you are), assigning inputs to subgroups will help make group changes quickly.

Next time: what goes into creating an effective mix.

 

Duke DeJong has more than 12 years of experience as a technical artist, trainer and collaborator for ministries. CCI Solutions is a leading source for AV and lighting equipment, also providing system design and contracting as well as acoustic consulting. Find out more here.

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Posted by Keith Clark on 03/06 at 12:46 PM
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Wednesday, March 05, 2014

Happy Mondays Celebrates 25-Year Anniversary Tour With Avid S3L

Front of house engineer Gerry Parchment and monitor engineer Jasen Hattams pushed to carry their own consoles

For a UK tour by British alternative band Happy Mondays celebrating the 25th anniversary of its second album entitled Bummed, the band and crew returned home when possible.

As a result, the tour made extensive use of splitter buses—van-based transports that double-up as both gear and personnel carriers. Although space on the splitters was limited, front of house engineer Gerry Parchment and monitor engineer Jasen Hattams pushed to carry their own consoles to ensure the most consistent sound possible throughout the tour.

For this they engaged Miles Hillyard at SSE Audio Group to supply two Avid S3L mixing systems.

Comprised of a compact control surface, HDX-powered processing engine running on-board AAX plug-ins, and networked remote stage boxes for up to 64 inputs, S3L’s modularity and easy Ethernet AVB connectivity over lightweight Cat-5e cables allowed Parchment and Hattams to carry this highly portable package in the back of the lighting crew’s van throughout the tour.

“Initially there was to be no control package at either end of the multicore, as we were in splitters and space was tight” says Parchment. “This left both Jay and myself with a myriad of in-house analogue and digital desks at the venues. [With Avid S3L] we could achieve a consistent sound and be ready to soundcheck the band in 30 minutes from the time the equipment got onto the stage,” Parchment continues. “We would not have been able to achieve this with any other system.”

“My Production Manager loves the footprint,” states Hattams. “It’s compact, and the less space we take up, the more seats are sold out front—that makes the band and promoter very happy. And I love a lampie carrying my desk.”

Both Parchment and Hattams are experienced engineers who have mixed on Avid live systems over the years. They both felt at home with S3L, as it runs on the common VENUE software platform and offers the same functionality as Avid’s larger consoles, including on-board plug-in processing,  integration with Pro Tools for show archiving and Virtual Soundchecking, and complete show file compatibility across all systems.

“[Avid] Profile has become the staple choice for many engineers,” Parchment notes. “[For the ‘Mondays tour] I initially took an old show file from a Profile system and modified it [on the S3L] as the tour went on.”

“We had a great show that [first] night,” adds Hattams. “The band was happy with the sound, so job done.

“On monitors I had 28 inputs, with 24 outputs doing a mixture of wedges and IEMs,” he continues. “There were 10 wedge mixes and four stereo IEMS. I have found the system easy to use, and most of all, it’s stable.”

A key feature of the is Virtual Soundcheck, which allows engineers to fine-tune mixes based on previous live recordings. With the combination of Pro Tools and Virtual Soundcheck, it’s possible to EQ the room, set up snapshots, and more—all without requiring the presence of the band.

“The single most impressive aspect of Virtual Soundcheck is its ease of use,” says Parchment. “That and the fact that a single channel may be switched between a return or live input. The ‘Mondays were present at sound check every day, but I used [Virtual Soundcheck] for setting individual items such as dynamics and effects.

“I recorded most of the Bummed album tour for archive and the intention is to mix certain shows for future use,” he concludes. “Recording [to Pro Tools] via a single Ethernet cable is heaven-sent.”

Avid

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Posted by Keith Clark on 03/05 at 04:26 PM
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Roland Systems Group Expands Digital Snake Offering With New S-2416

Provides a 24-input x 16-output analog and 8-input x 8-output digital for a total of 32 input and 24 output channels

Roland Systems Group has announced the S-2416 Stage Unit, a new digital snake stage unit offering a 24-input x 16-output analog and 8-input x 8-output digital for a total of 32 input and 24 output channels.

In addition to the analog and digital I/O, the S-2416 has two REAC ports enabling the ability to cascade an additional snake for expanded I/O or for a fully redundant, zero-loss audio back-up solution. Further, the discrete mic preamps are newly developed to enhance sonic quality.

The ruggedized chassis is an EIA design with a 4RU size, designed for touring and fixed installation. The analog connectors employ rugged Neutrik connectors and the inputs have 3-color indicator lights that provide phantom power, clipping, and signal presence at a glance.

The AES/EBU ports on the rear are 25-pin D-sub types that enable other digital audio devices such as loudspeaker processors, amplifiers and digital devices to be directly connected.

The S-2416 is the first digital snake capable of a cascade connection, making it possible to expand the I/O up to 40 x 40 channels. A simple activation of a switch to Cascade Mode allows a connection to an additional REAC digital snake. The cascade feature allows for more complex and flexible V-Mixing System configurations.

The S-2416 supports 24-bit 96 kHz, 48 kHz, or 44.1 kHz when switched to clock master mode. Word clock input and output enable a master clock signal to be connected when using AES/EBU.

Mic preamps can be remotely controlled using any V-Mixer, R-1000 or S-4000R connected by RS-232C or by using the S-4000 RCS remote-control software on a computer (Mac or Windows) connected via USB.

Fundamental to S-2416 connectivity is REAC – Roland’s Ethernet Audio Communication protocol for low-latency, high-quality digital audio transfer. Products using REAC technology are installed today in many venues and have been used in high-profile events worldwide.

REAC’s 24-bit, 40- x 40-channel protocol delivers pristine digital audio via lightweight, inexpensive and easy to install cable (Cat-5e/6). REAC technology is immune to externally induced signal quality degradation or hums and buzzes typically found in analog systems which leads to freedom in cable placement resulting in lower cost and better sound.

The REAC family of products is varied and includes digital mixing consoles, digital snakes, personal mixers and multi-channel playback and recording, all of which work together in a plug-and-play fashion.

The new S-2416 is expected to be available in Quarter 2 of 2014.

Roland Systems Group

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Posted by Keith Clark on 03/05 at 10:12 AM
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Allen & Heath iLive & ME Personal Monitoring On European Tour With Ane Brun (Inlcudes Video)

iLive-T112 system handles both FOH and monitors,

Swedish PA company Parashoot supported the recent European tour by Norwegian singer-songwriter Ane Brun with an Allen & Heath iLive digital mixing system as well as A&H ME personal monitoring systems.

Specifically, Parashoot deployed an iLive-T112 system to handle both FOH and monitors, joined by xDR-16 I/O expander racks positioned stage left and stage right, along with an iDR-16 MixRack.

Parashoot, which has worked with Ane Brun since 2007, served a line-up comprised of two drum kits with open miking, an upright bass with FX pedals, various guitars, two keyboards, backing vocalists, and Brun.

All musicians except one were on in-ear monitors working with ME personal mixing systems. Parashoot provided five ME-1 personal mixers, with an ME-U hub delivering power and distribution on stage.

“I took the ME-1s down to rehearsals and everyone was thrilled with them,” notes Parashoot sound engineer and owner Oscar Söderlund. “The layout is easy, you can pan and it’s a sleek looking tool.”

As Söderlund was also managing monitors from FOH, the ME-1s helped take the stress out of soundcheck and the show so he could focus on the PA and Brun’s mix. He discovered Allen & Heath in 2008 when a friend suggested he try iLive, which resulted in him investing in an iDR-48 MixRack and iLive-T112 system. The company now owns a flexible hire stock, including three iLive-T112 surfaces, two iLive-R72 surfaces and four MixRacks, which are interchangeable systems.

“When I first toured with iLive, the feeling was great,” he states. :You pull up the fader and it sounds so different to what I had heard before with other consoles, it’s so natural - I fell in love.  What struck me was the FX, they are the best emulations in the digital console business. I was dealing with 28-inch bass drums with no padding, and I found I could really extend the reverbs in my mix and create great landscapes.”

“It’s not often that you do something and think I couldn’t have done that any better. Would it have been better with another console? No, I’d always go back to A&H. It’s been the best five years of my life,” he concludes.

 


Allen & Heath

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Posted by Keith Clark on 03/05 at 09:53 AM
Live SoundNewsConcertConsolesDigitalEngineerInterconnectMixerMonitoringSound ReinforcementPermalink

Tuesday, March 04, 2014

Austrian Mixer Lukas Zilka Takes Soundcraft Si Expression 3 Console To The Big Stage

Upgrade will assist with his own productions as well as with the services he routinely offers to larger companies

Lukas Zilka, a freelance mixer for live concerts, press conferences, corporate events and national balls, recently added a Soundcraft Si Expression 3 mixing console to the inventory of his company, Live Productions–Event Services, based in Purkersdorf, Austria.

The console upgrade will assist him not only with his own productions, but also with the services he routinely offers to larger companies.

Zilka manages audio for some of the nation’s top balls, including the “Bonobonball” (Candy Ball) in Vienna. His previous experience with the Soundcraft GB8 analog console installed at that venue is what led him to grow his business with the Si Expression 3.

“Simply put, the value of the Si Expression 3 is unmatched, as other brands cost an arm and a leg for the same feature set that Soundcraft offers,” Zilka says. “I mix children’s plays for one of my regular clients, which is a local community church, and as the performance group expanded over the years and became increasingly professional, the small racks on the old Yamaha console couldn’t cut it anymore. I needed more outputs and more faders, so when I saw the Si Expression 3, I knew my search had come to an end.”

The Soundcraft Si Expression 3 offers 30+2 faders along with flexible onboard and expansion I/O options so that every one of the 66 input processing channels can be used. The console also includes the EMMA DSP processor as well as built-in FX engines that do not impact the mix regardless of load. The fader groups are illuminated with FaderGlow, which color-codes the motorized faders for an intuitive mixing experience.

“So far, I’m loving the digital experience,” Zilka notes. “I really like the easy handle on all channel parameters, as there is no need to jump into menus to get the effects I want. When I did a charity concert, I noticed the advantage of storable pre-amps, which was not present on my old mixer. I also enjoy using the ViSi app on my iPad for corporate events and press conferences.”

“Overall, Soundcraft delivered on every aspect of its name, providing me with an experience that cannot be found elsewhere for this price,” Zilka concludes. “The Si Expression does a perfect job and helps me create the exact sound that I want. Its ease-of-use features such as FaderGlow are revolutionary for this industry, which contains too many cluttered mechanics from other brands.”

Soundcraft
Harman Professional

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Posted by Keith Clark on 03/04 at 02:43 PM
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Penteo 4 Pro Stereo-To-Surround Up-Mixer Plug-In Now Available

AAX64 support, new features and a 33 percent promotion discount

Audiotech Digital has announced the release of the new Panteo 4 Pro plug-in that discretely converts stereo to 5.1, offering exacting control over sound image placement to create high-quality, sonic-free and natural-sounding surround. 

Penteo 4 Pro is 100-percent ITU down-mix compatible to the original stereo. The plug-in offers an intuitive visual interface modeled after vintage gear and six automated preset modes as well as advanced manual controls for fine-tuning.

Mike Minkler, multiple Oscar-winning re-recording mixer at TODD-SOUNDELUX, explains, “I’ve used Penteo on almost every project for five years now, such as Django Unchained, Twilight Saga, and Inglourious Basterds. The new plug-in really shines. With a flick of a switch I can take a two-channel mix and get a quality 5.1 mix, which is a huge asset in these budget-conscious times, especially when independent films just can’t afford more than a stereo source for me to work with.”

Native AAX64 Support
Penteo 4 Pro now natively supports Avid Audio eXtension (AAX) for true 64 bit processing. As a result, Penteo 4 Pro AAX64 uses 52 percent less CPU processing power thereby allowing a sound designer more flexibility to simultaneously use more tracks in a mix than ever before. 

“Penteo produces such a warm surround mix, and we are thrilled that now Penteo supports Avid’s AAX64 format,” says Ed Gray, director, Avid Partnering Programs. “Now sound designers will benefit by being able to run more instances and tap all the new features of ProTools 11.”

New Features

—Touch Screen Controls - In addition to mouse operated controls, Penteo 4 Pro is the first up-mixer to support multi-touch controls. Mixing is not a linear, sequential process—Penteo’s unique multi-touch interface allows a sound designer to control more than one Penteo feature at once, thereby creating a natural mixing experience through simple hand contact.  Penteo 4 Pro is the world’s first multi-touch plug-in designed for use with the Raven MTX Surround console. 

—Pro Tools Automation Keyboard Shortcuts for all parameters - Quickly add and select parameters for automation using standard control/alt/cmd shortcuts.

—Usage Preferences - With choice of circular or vertical knob drag usage options, a sound designer can select the most natural feeling knob movement for mouse or multi-touch.

—New bypass channel routing - Stereo bypass channels now routes to the ProTools channel output standard.

To celebrate Penteo’s AAX release, from March 4 to April 16, 2014, new purchases of any format of Penteo will receive 33% off the purchase price.  Free trials are available here.

Audiotech Digital/Panteo

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Posted by Keith Clark on 03/04 at 02:31 PM
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