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Monday, September 27, 2010
Digital Consoles: 10 Reasons Why You Can’t Avoid Purchasing Digital
Modern event production is growing used to the benefits of digital mixers, and here's why
It’s not surprising that analog consoles remain in use because of the hat trick that they’re paid for, they’re in place, and operators are used to working with them. Used analog desks also cost a fraction of their original price.
However, modern event production is growing used to the benefits of digital mixers. Here are 10 reasons why digital consoles win the day in today’s productions.
In the digital world – where outboard dynamics, effects and EQ are built into the desk, and loudspeaker processing and amplifiers are being combined with, or built into, loudspeakers – the mixing board is the easier of the two to replace, while the loudspeaker system can be incrementally upgraded one zone, speaker, amp or processor at a time.
Many artists’ engineers travel with their own desk, even when the entire sound system isn’t being carried, as it provides consistency and comfort. The ability to quickly have a show ready, with only loudspeakers needing to be EQ’ed before soundcheck, means load-in can start later, freeing time for other appearances and travel the day of show.
Those that don’t travel with their own console can email a file from a previous appearance with that make of digital mixer, or build a file using the manufacturer’s offline editor. Just using a console’s editor allows engineers to familiarize themselves with a new desk.
Multi-band shows can be as effortless as peeling off stacked backline and moving a few XLRs, which takes longer than recalling the correct console scene. “Charting” consoles or using separate inputs becomes unnecessary. When sharing inputs, the next act’s settings are recalled faster than the previous band’s gear can be carted off the stage.
Saving scenes for acts or events that repeat regularly, especially when there are multiple events in a day, makes it easy to quickly set up and get soundchecked when time is short, perhaps even after the support act or the wedding. Acts that return regularly can afford to skip soundcheck, allowing them to arrive later in the day, a great benefit when they must travel long distances.
Custom input libraries for a venue’s microphone inventory speed and simplify programming. Writing and storing specific settings for the house mics allows sound check to get started in a few minutes by inserting the right files in the corresponding channels, even by a less experienced engineer.
Application-specific settings for dynamics and effects also make it easier to program a digital console. Meaningful, logical names for library settings – like “Kick Gate,” “Vocal Comp” and “Snare Verb” – can make it simple to find and load a file for a gate, compressor or reverb that is pre-tweaked for a particular application.
Output EQ for specific combinations of house loudspeakers and vocal mics (and even music style) can make the chore of tuning the mains and monitors much easier.
Each vocal mic and wedge combination benefits from particular output EQ settings. The same vocal mic with a double wedge would require more drastic cuts than a single wedge.
Output EQ for the main loudspeakers often needs to be adjusted for different types of music, especially for different levels of SPL. Settings like “Double Wedge 58” or “Reggae Mains” can save time before every show.
The great asset of early digital effects was their powerful library of presets. Though most desks have libraries, one common shortcoming is a lack of meaningful scenes that can be quickly recalled.
Generic “festival” scenes for different types of bands can make it easy to get a band’s sound check started, rather than starting from scratch with a zeroed desk.
Just having the inputs named, initial input gains, EQ, dynamics and effects loaded saves time, and the time spent tweaking those settings can result in a better mix. A simple four-piece of drums, bass, guitar and keyboard setup with a vocal input for each musician could save an hour.
Templates for particular events, whether they be panel discussions, praise band rehearsal, karaoke or wedding receptions can relieve your lead engineer from having to be at every function in the building.
The ability of every staff member to call up a preset and get a couple mics and a playback source to work can help better utilize a facility outside of weekend “prime time” hours.
Security features that lock out certain functions can keep a console from being intimidating or dangerous in the hands of novice engineers. Remote monitoring or control of console by a technical manager or outside vendor can help with troubleshooting in the case of an emergency, perhaps saving an on-site call.
As the benefits pile up, it seems inevitable that your next purchase could be a digital console. So let’s look at some of the newest offerings via the PSW Photo Gallery Tour of the latest digital consoles on the market.
Mark Frink is Associate Editor Live Sound International.
Phase Alignment Between Subwoofers & Mid-High Cabinets The Series: Part II
In this second part of our series from D.A.S. Audio, we take a look at coherence curve and the first of several illustrative examples.
In the first part of our series, we took a look at what exactly phase and polarity are, what causes phase to vary, and what is meant by phase alignment.
Coherence Curve
The coherence curve that Fast Fourier Transform based measurement systems provides indicates the probability that the measurement is reliable.
It’s very common to find a coherence curve (ranging from 0 to 1, or from 0% to 100%, depending on the measurement system) with low values in part of the spectrum.
We should not trust the magnitude or phase frequency responses for those bands for which our measurement system shows low coherence.
There are two main reasons for poor coherence:
1) Reference signal is badly synchronized with the measured signal.
We can test for this easily if we initiate a measurement without having first synchronized the measurement signal using the “Delay Finder” on SATLive or the equivalent function in other systems.
In this case it will be seen that coherence for the high frequencies is very low, as seen in Figure 6.
Figure 6: SATLive’s coherence scale can be seen on the right-hand side, ranging from 0 to 1. This curve’s coherence trace has been stored and loaded with the Trace Manager utility, and is shown as a thinner blue trace.
2) Reflections.
These will cause coherence in some frequency bands to be low. We should not trust measurements on those bands.
If we are interested in measuring a part of the spectrum that has poor coherence, we can change the microphone location.
When it comes to adjusting phase, we should check the coherence in order to know which parts of the measurement are reliable and which ones are contaminated by reflections, reverberation, etc.
Example 1: “Scaled down measurements: the subwoofer and the mid-high box share a frequency band”
Before trying to do these adjustments for the first time in a real-life situation, where one may not always have enough time and where conditions are far from ideal, scaled down measurements can be handy to get some practice with the procedure.
We’ll assume that you already know how to take transfer function measurements with the measurement system you are using, and that the equipment used is adequate.
To synchronize the reference signal to the measurement signal, we need to measure the system’s impulse response, which is better obtained from the high frequency band, and therefore we’ll always use the mid-high units for synchronization.
There will be times when we’ll need to shift the subwoofers backwards by adding delay, and times when we will need to move them “forward” with “negative delay”.
Since such thing does not exist, we’ll add an initial time delay that is the same for all bands, so that we will be able to add or subtract delay from the subwoofers’ initial delay time. Once the system has been adjusted, we’ll get rid of excess delay, as will be seen in the examples.
Let’s now run a scaled down measurement of an 18” subwoofer and a mid-high unit.
Subwoofer cut-off frequencies for the real system will be:
HPF LR24dB/Oct, 30Hz
LPF LR24dB/Oct, 85Hz
Cut-off frequencies for the real mid-high unit will be:
HPF LR24dB/Oct, 50Hz
LPF LR24dB/Oct, 20KHz
We’ll use two 4” speakers to get some practice with the phase adjustment procedure.
For our 4” to behave acoustically like our real, full-scale systems, we will need to scale the crossover frequencies up.
To do that we’ll multiply the real system cut-off frequencies by the ratio of the real system to our scaled down box, i.e., we will multiply the cut-off frequencies by 18”/4” = 4.5.
Therefore the cut-off frequencies for the scaled down measurements, which will be entered in the processor for the 4” system, will be as follows:
Cut-off frequencies for the scaled down subwoofer system will be:
HPF LR24dB/Oct, 30Hz x 4.5 = 135Hz
LPF LR24dB/Oct, 85Hz x 4.5 = 382Hz
The cut-off frequencies for the scaled down mid-high system will be:
HPF LR24dB/Oct, 50Hz x 4.5 = 225Hz
LPF LR24dB/Oct, 20KHz
We’ll leave the low pass filter for the mid-high box at 20 kHz. Otherwise we would be in the ultrasonic range.
For these measurements we used a DAS Arco 4 enclosure as the subwoofer, lying on its side.
The box used as a mid-high, also a DAS Arco 4, is placed somewhat higher up, and some 15cm (6”) behind the box being used as a subwoofer, as shown in Figure 7.
The microphone is placed on the ground, at 90cm (3 feet) from the simulated subwoofer.
Figure 7: Side view set-up for the scaled down measurements used for examples 1 and 2.
In order to notice more easily the difference between aligning the phases or not aligning them, it is recommended to set the acoustic levels of the mid-high and the subwoofer the same in the band being shared, 225Hz to 382Hz in our exercise.
The procedure is as follows:
1) Enter 20 ms as the delay time for each of the outputs in the processor (This is an arbitrary value; a different delay time can be used).
2) Let’s first work just with the mid-high. We’ll use the “Delay Finder” utility to add the required delay to the channel with the reference signal, i.e., to synchronise the reference signal to the measured signal. (See the user’s manual for SATlive or your analysis software for more information).
Figure 8: This is the magnitude frequency response curve we are trying to improve on. A cancellation can be seen around 400Hz, and therefore within the frequency band being reproduced by both boxes.
3) Measure the magnitude frequency response for the complete system before doing phase adjustments. At worst, we will see significant cancellation in the frequency range being shared by both enclosures. The measurement can be seen in Figure 8.
4) Mute the subwoofer output, and un-mute the mid-high output in the processor.
5) Measure the mid-highs and save the curve. In our example, the curve in Figure 9 is obtained.
Figure 9: Magnitude and phase frequency response for the mid-highs.
6) Mute the mid-highs and un-mute the subwoofer output.
7) Do not use the “Delay Finder” again!!! (i.e., do not synchronize the reference signal to the measured signal again).
Remember that we are comparing phase on both outputs, i.e. we are measuring the difference in time arrival between the two signals as a function of frequency.
Therefore the synchronization delay for the reference signal should not be changed on the measurement software again. Keep in mind that we took the mid-high box as our timing reference because it is the signal from which the best impulse response can be obtained.
8) Measure the subwoofer and compare the phase curve with that of the mid-high box. The result can be seen in Figure 10.
9) Add or subtract delay from the subwoofer output until the two phase curves overlap around the crossover frequency. Do not forget to save the curves.
The curve with the steepest slope of the two is the one with the most delay. Therefore, it seems clear in this case that we will have to subtract delay from the green curve, i.e. the subwoofer output.
We’ll be able to do this because we initially added a delay of 20ms to the two outputs.
Remove some of the delay from the subwoofer output and the green curve will loose slope and shift upwards, and the two phase traces will overlap within a fairly wide band.
Figure 10: Capture shows the difference in phase between the subwoofer and the top box for the frequency band being shared (160Hz to 400Hz). This explains the cancellation seen near 400Hz, and the fact that the level in the rest of this shared band does not increase significantly.
The delay on the subwoofer output ended up at 18.666ms. From 150Hz to 400Hz the two curves overlap, i.e. they are in phase within the entire band they share.
Therefore, if we compare two phase curves and we want to minimize the difference in phase between them, we need to remember the following: if a curve has a steeper slope than the other, it’s arriving late and we need to take away delay.
If a curve has a gentler slope than the other, it’s arriving early and we need to add delay.
Keep in mind that, in our example, the subwoofer was physically forward with respect to the top box, so we could have mistakenly assumed it was the subwoofer that needed delaying.
Figure 11: Top box and subwoofer responses with phase adjustment. It can be seen that phase overlaps in the shared frequency band, which means they will be summing perfectly in phase.
Do not forget that filters have an effect on phase, and therefore we cannot predict if we need to add or remove delay until we see the measurements.
Let’s see what would have happened if we had increased the delay time to the subwoofer instead of reducing it.
In Figure 12 delay has been added to the subwoofer output until the most overlap was obtained. The subwoofer delay ended up as 22.276ms.
Phase overlaps in the 250-300 region, which is very little. Below 250Hz the blue phase trace is below the green one, whereas above 300Hz the green curve is below the blue one; i.e. there’s phase difference between them.
Figure 12: In this specific case, delaying the subwoofer does not make the phase traces overlap for the entire band.
10) Measure the system frequency response and compare it to the initial measurement.
If phase has been correctly adjusted, subwoofers and mid-highs will sum in phase and this will be reflected on the magnitude frequency response.
Figure 13 compares the system combination without adjustment (red trace), with 22.2766ms delay on the subwoofer (green trace) and with 18.666ms (blue trace).
Figure 13: In this particular case, the subwoofer and the mid-high box sum optimally when we take away delay from the subwoofer.
It can be clearly seen that the best sum occurs for the 18.666ms subwoofer delay.
11) Take the lowest delay value and subtract it from the subwoofer and mid-high so that at least one of the outputs has a delay time of 0ms.
At this time there’s a delay of 20ms on the mid-high and 18.666ms on the subwoofer.
Since we had added 20ms just as an arbitrary amount to be able to add or subtract from that as needed, once the adjustments have been made we no longer need that excess delay: subtract the lowest delay time from the two outputs so that one of them has 0ms.
In our example the mid-high output will end up with 20ms – 18.666ms = 1.334ms. The subwoofer output will have 18.666ms – 18.666ms = 0ms.
Stay tuned for the coming articles in this series, where we’re lay out additional examples. Want to get a jump on the reading? Head on over to the DAS Audio Website where you can DAS Audio Engineering Department.
The new bundle brings the power of Waves plug-ins to the video edit suite.
Waves has announced the introduction of the all new Video Sound Suite.
First impressions are lasting impressions, and with producers and clients focused on every frame, even rough cuts have to win them over.
That means when you hit play, what comes out of your speakers has to be every bit as good as what’s up on the screen.
With the new Waves Video Sound Suite of audio plugins, now your location sound can instantly sound as good as your video, quickly and easily.
Using the same plugins as the industry’s leading movie and game audio professionals, video professionals can do it all: reduce noise, clean up and enhance dialog, smooth out and maximize volume, re-create room acoustics and more.
Video Sound Suite integrates seamlessly into Avid Media Composer 5, Apple Soundtrack Pro, and Sony Vegas.
Because they’re real-time plugins, it’s no longer necessary to render or create new files for every audio adjustment, and making audio changes weeks or months later is a breeze.
Video Sound Suite includes: Renaissance Compressor: Helps keep volume levels under control, for smoother, more consistent cuts.
IR-L Reverb: Lets you place sounds in real ambient spaces, add atmosphere, and smooth out tight edits.
DeEsser: Tames sibilance, the ‘ess’ and ‘shh’ sounds which can make voices sound harsh and distorted.
W43 Noise Reduction Plugin: Reduces ambient noise like hiss, hum, traffic, wind, and air conditioning.
Q10 Equalizer: Lets you enhance frequencies, cut lows & add highs so the voice cuts through, or zoom in and clean up problem areas.
Midas PRO6 The Choice Of Riverside At Poland’s Metal Hammer Festival
FOH engineer Schindler chose the PRO6 for its quality sound and powerful automation.
The leading Polish rock band Riverside chose a Midas PRO6 live audio system when they headlined the Metal Hammer 2010 Festival in Katowice, Poland alongside an array of acts including the multi-Grammy award winning Korn.
Audio equipment for the entire festival was provided by Kraków-based rental company Prosound.
FOH engineer Daniel Schindler first encountered the PRO6 touring as a monitor engineer for Turkish singer-songwriter Nil Karaibrahimgil.
“From the first rehearsal for the Nil tour, I was blown away by the PRO6 sound,” he says.
“It sounds like an XL4 without 120m of copper between the source and the board. I’ve been looking forward to using the console with Riverside ever since. I also trained with German Midas distributor Mega Audio, and am proud to be a Certified Midas Digital User.”
Schindler praised the natural sounding preamps and onboard dynamics and FX processing as standout features, as well as the console’s design. “It feels like using the analogue Midas equipment we all love so much.”
“The way of working is intuitive and a step towards the future. Once you’ve worked on it, you start missing that on other consoles.”
Schindler used the PRO6’s pitch shift feature to add second and third sets of vocals to the live mix, supplementing lead (and sole) vocalist Mariusz and recreating the harmonies captured on recordings in a live environment.
While enjoying the analog sound of the console, Schindler is quick to appreciate the digital advantages of the PRO6. “Riverside are quite complex to mix,” he explains.
“Sometimes you need to change the full live mix in just two bars. But the PRO6’s powerful automation makes this simple. The POP(ulation) Groups totally changed my way of working, in a positive way.”
“It makes the access to the right channels even faster than on a big analogue console and gives a great overview of the really important channels. I also like that the VCAs give access to the assigned channels in the same way as the POP Groups, while the Area B is great for channels that always need to be accessed directly.”
Schindler reports the Midas PRO6 also proved its worth in extreme conditions. The powerful screens are clearly visible, even in bright sunshine; “a good thing if you have to soundcheck in Turkey at 2pm under blue sky in 48°,” he reports.
Even a thunderstorm failed to faze the PRO6. “I had to dry it out completely after 5cm of water got in the back of the case; the console continued to work like nothing ever happened.”
Schindler is now looking forward to more opportunities to use the PRO6. “On Polish tours we always use equipment from Fotis Sound in Poznan and they already own a PRO6, so the future’s looking good,” he concludes.
Obviously, training is necessary for a worship tech team, however with whom does the responsibility lie?
One of the most discussed topics in the world of church sound and A/V is training.
Integrators are installing more sophisticated and complex systems as many churches are incorporating A/V and production into their weekly services.
Churches spending big dollars on these installs are expecting big results.
The problem rears it ugly head as volunteers will little or no training try to operate these highly technical systems.
Whose Problem is it?
The technical advancements that have happened in the church market over the last 20 years have been astounding.
The average church has gone for a 6-8 channel mixer to 32 channels and from maybe 1 wireless lav to systems of 8 or more. The most impacting and profound change that have driven this have been the move from piano, organ and a single “minister of music” leading worship to contemporary worship.
Today at a typical worship service it is common to find a worship team consisting of 4 or more vocalists, worship leader, guitars (electric, acoustic and bass), keyboards, piano, and drums. This could also include a flute player, violinist, and even a brass section.
Yes the typical worship has changed. Now also add in to this the drama that takes place with 4 -5 actors and numerous sound effects. At time church looks and feels like a cross between Broadway, touring concert and motivational seminar.
I am not going to go into on a dialogue as to what is proper and appropriate. That is for each congregation to decide and choose based on their mission and calling.
The point that is being made is that in a short 20 years the church now has an expectation placed on it to provide relevant, entertaining, uplifting preplanned and excellent programming.
The A/V industry has not necessarily had a great track record on training the end-user. In fact it does not even have a great track record of training its own people. Up until recent years and the attempt at NICET certification and now C-EST training the industry was primarily made up of home-grown, self-taught personnel.
I would even argue that it is still that way as many integrators do not take advantage of the training offered by Industry associations like NSCA.
In addition often the integrators that are installing systems into churches that are doing production have had no experience in what the demands of a production orientated church service is.
A friend of mine who has been a worship leader at churches that are very production orientated and have had millions of dollars of technical equipment stated “it (mixing for church) is not like mixing for a 2 hour rock concert.”
“You have a worship team, band, sometimes orchestra, the spoken word, video elements, lots of transitions, drama, loud moments, quiet moments, even silence….. and then to top it off you have unpaid volunteers who are the talent on the stage.”
I would add to that in most cases you also have an unpaid volunteer at the mix position.
I find it astounding that a church has no problem investing $250,000 sometimes over $500,000 on A/V systems (sound, theatrical lighting and video) and does not even consider hiring some one to run and maintain it.
Every church that I have had the opportunity to work with has had a least one paid music person on staff.
Often there was a whole department that included worship leaders, orchestra director, choir directors and producers.
Yet these churches who pride themselves on the quality of the music and production can’t even find in their bloated budget enough money to hire a quality person that will be the conduit to reproducing and pulling together the quality music and production.
How can it be solved?
It can be easy to sit back and point out a problem, but it gets a bit more difficult when you have to put together a solution to the problem.
Before stating a solution a plethora of questions can be asked;
• Why are the Christian Colleges and music schools not offering degrees in technical ministry?
• Why are there not more vocational schools that offer training?
• Why does the A/V industry not offer end-user training?
• Why is the not a training school like the recording industry has?
• Why? Or Why ask Why?
The technical knowledge deficit that exists today is a result of many factors that include the rapidity that the church has adopted technology. An ignorance or misunderstanding of what it requires to operate and maintain technical systems.
Integrators that have sold systems claiming the are easy to operate and then provided minimal or no training. Volunteers at the church turn over often, an operator who was trained is no longer involved and did not pass the knowledge on to others or passed on incorrect or confusing information.
Also, the church is filled with “experts”, the guy who works for the cable TV company as an installer so therefore is an authority on church sound and is too arrogant to seek training or input for qualified individuals.
I have come to the cynical conclusion that most churches are controlled by the church secretary, custodian, and/ or sound operator. For some reason a majority of churches seem to let this select group of individual dictate church policy and decision making.
What is needed to fill the gap? First of all churches must take responsibility and be willing to fund and purchase training resources.
This could include hosting seminars, paying experts to come on site to do training, paying for technicians to go to seminars or purchasing books and materials that are relevant.
This is common sense stuff; would you buy a car if you did not know how to drive? So why do so many churches purchase technical systems with no clue how to operate them.
Secondly, system integrators need to step in a provide complete comprehensive training.
If the integrator is qualified to install the system then the integrator should be responsible for training on the system.
One of my favorite arguments for design/build contracting is that if the integrator properly designs the system, properly installs the system and then properly trains the end-user, whose problem is it if the system does not work?
In my opinion an integrator in a design build scenario should offer at a minimum a 1 year no fault warranty on a system. Baring the proverbial “act of God” there should be no labor or equipment charge to remedy any problem that arises, even if it is operator error. After all it was the integrator who trained the users.
Thirdly, the industry and its associations should seek avenues to provide end-user education and training.
An educated end-user makes wiser choices in selecting integrators and equipment resulting in better installations. In addition a trained end-user would serve to raise the over all sound quality and consistency that the general public experiences.
Who is going to take the lead?
We all like to point out problems but who is going to step up and offer solutions? Perhaps it will come through the church.
Networks of churches, whether denomination or style based, are already in existence. It maybe that groups will form and churches will train other churches. Maybe the industry will formulate sanctioned end-user training through one of the associations, like the C-EST training offered to installers.
Most likely a passion driven integrator or Mega Church technical director will develop a curriculum that is web based or video based that will serve to educate system operators.
The well know saying “it’s the service after the sale” that matters applies well here. If the church is the customer then it is time for the integrator and the industry to stand up and service the daylight out of the customer!
Finally, before signing off, I should also mention that there exist resources like the How to Sound workshops that the end-user can bring to their area to provide comprehensive training, which currently help to bridge the gap in this education divide.
Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 25 years.
The XE Series is a small, efficient unit intended for portable and fixed installations.
PKN Controls Ltd. has announced the introduction of the XE Series professional amplifiers at PLASA 2010.
Designed for portable and fixed installations, the XE Series includes the XE2500, XE4000, XE6000 models.
The PKNC XE Series construction is based upon the latest in high frequency energy conversion technology, making these solid state devices extremely reliable.
All models within the newly announced XE line feature graphic LCD displays and two JOG dials to easily manipulate system parameters.
The devices include internal web servers for remote access via either internet browser (JAVA based), or with PKNC control software.
Additionally, embedded microcontrollers protect the amplifier against any kind of damage in real-world applications, maintaining the functionality of amplifier.
Features:
• High output power and high efficiency
• Advanced high frequency switching technology
• Small size, low weight, ideal for touring applications
• Networkable amplifier over standard Ethernet
• AMPControl software and JAVA based control
• All functions can be accessible by remote
• Graphics LCD for easy setup and diagnostics
• Two JOG button for fast navigating
• Digital volume and input sensitivity control
• Built-in programmable limiter
• Output Peak and average Voltage,Current measuring
• Displays actual load impedance
• Embedded web server for set&view parameters by a web browser
• Five user profiles
• Intelligent protective functions
• High frequency resonant power supply with active PFC function
• Wide mains voltage operation range
Q: “Like many people, I use a maximizer (the Waves L1, if it matters) on my mixes prior to burning them on CD.
Many of my clients prefer their mixes as loud as possible even though I sometimes hate it.
I sometimes find it gets really grainy or gritty sounding when too much of this process is applied.
What can I do to get better sound, but keep things really loud?
A: You’re certainly wise to be cognizant of the amount of processing you’re using, as there has been somewhat of a backlash among audio engineers against overuse of maximizers in years past.
Many of us now fight like dogs to keep the dynamic range in our material and sell our clients hard on how this is really better and more musical.
Of course, at the end of the day you have to apply the axiom that the customer is always right and try to do what they want.
To speak directly to your situation, the L1 acts mainly on the peaks of your material, and it does do a good job of making complex music signals sound a lot louder and more energetic with reasonably little coloration.
As it is. this product practically built the project mastering industry all by itself.
However, like any creative tool you can push it beyond its intended boundaries and make it sound bad.
The gritty quality you refer to is probably a function of too many waveform peaks in your material being compromised.
In order to get you the headroom it needs to normalize or turn up the whole program, it has to trim back the peaks. Change too many of these peaks too much and the sonic character of the material begins to change.
The solution? Aside from the obvious, “back it off,” you can employ more common forms of compression to reduce dynamic range before you apply your maximizer.
A good multiband compressor applied to the mix can reduce dynamic range in a more gentle way (in terms of peak waveform distortion). You can dial in more compression on a range of particularly problematic frequencies thereby opening up more headroom for the rest of the signal.
Applying RMS (as opposed to peak style) compression of any type is likely to help you. The ability to tailor it to specific frequencies is just another way to further fine tune the process.
Once you do this you can then apply your maximizer to what’s left and potentially get better results.
There will still be peaks to act on that will allow your maximizer to get your overall mix a few dB louder, but it will not have to hit them as hard if the material has been somewhat compressed ahead of time.
Your results will vary depending upon the material and the exact techniques you employ. Compress any signal too much and you will end up with something resembling noise. There’s only so much you can do.
Also keep in mind that all compressors add some type of coloration to the overall sound, especially when they are used in excess. Many people find the sound of some compressors pleasing, while they don’t like others. You may have to try a few to find one you really like.
As always, we welcome input from the PSW community and would love to know your thoughts on maximizer use. Feel free to let us know in the comments below.
StagePro Deploys JBL VerTec Line Arrays For Two Colorado Festivals
Battling harsh elements, JBL VerTec was the clear choice for Colorado’s Country Jam and Rock Jam Festivals.
Festival production specialist StagePro were responsible for the production of two highly anticipated concerts in one of the country’s most difficult audio environments.
They successfully conquered high elevation, harsh winds and zero humidity with a complete JBL VerTec line array solution for this year’s Rock Jam and Country Jam festivals, staged in Grand Junction, Colorado.
With artists ranging from Alice Cooper and Cinderella to Keith Urban and Sawyer Brown, the events presented many of the nation’s top current touring acts.
Utilizing 48 JBL VT4889 fullsize line array elements and 32 VT4880A fullsize arrayable subwoofers, StagePro president Jay Waller and system tech Nick Mourn overcame the environmental obstacles plaguing the country and rock festivals for nearly 20 years.
“We originally purchased the JBL VerTec rig to help solve the sound issues related to the stage’s environmental surroundings and to provide a system that could withstand the harsh weather conditions in Grand Junction,” stated Waller.
“There is so much inconsistency with the atmosphere, hour to hour, that our sound check in the morning would not provide us with the levels we required for afternoon or night performances.”
“Before we had the VerTec line arrays, the mix engineers weren’t able to get the sound to travel as well through the thin, high-altitude air, which didn’t please the artists, fans or organizers.”
The VerTec rig now used by StagePro is well known by the engineers and artists who grace the stages of Country and Rock Jam, in their 18th and seventh years, respectively.
Acording to Walker his company’s VerTec rig is consistent with the rider demands of many top-tier touring groups, so there is no need to re-adjust the system for each artist.
“They use the same rigs in stadiums and arenas that we use for our festival stages, so bands can just leave their touring sound equipment in the truck, making everyone’s hectic schedules a little less complicated,” Waller said.
“The VerTec VT4880A subs are unbelievable,” continued Waller.
“Every time we set up a rig for these events, the engineers always walk in thinking we need more subs, but the 4880A truly offers more punch per pound, taking up less room and cutting costs. After listening, they all agree that it is an incredible loudspeaker.”
EAW Loudspeakers Selected For Malaysia’s The Opera Nightclub
The owners of the Opera were very unhappy with the previous system installed and system integrator Bina Teguh was specifically asked to install an EAW system.
The owners of The Opera, a nightclub located in Petaling Jaya, Malaysia, recently turned to EAW for a complete sound system upgrade.
Regional EAW distributor ProAktiv Systems supplied the EAW loudspeaker systems and Powersoft amplification, which is now installed in the main theater and VIP areas throughout The Opera.
The clubbing experience at The Opera is complimented by daily “Cirque Nouveau” performances featuring acrobatic and circus troupes high above the main floor in the three-story, 10,000-square-foot theater.
The main stage regularly features performances by dance, gymnastic and martial artists. In addition, the club frequently showcases recording artists, bands, DJs and video DJs.
The main audio system in the theater, which is modeled after grand European opera houses, comprises four EAW MK5396 two-way full range compact installation loudspeakers.
The four-inch voice coil, 15-inch drivers are complemented by four SB1002 dual-18-inch high-output subwoofers installed at floor level.
Two MK2326 compact installation loudspeakers and a pair of VRS18 subwoofers supply under-balcony fill and delay. The main EAW system is powered by a combination of Powersoft K3 and K8 amplifiers with K2 and KF3 model amplifiers for the fill system.
EAW MK2396 loudspeakers, supplemented with VRS18 subwoofers, provide coverage in the luxury VIP area, which offers a commanding view of the main floor and stage. Four EAW compact two-way JFX290i speakers provide in-fill while SMS5 speakers deliver coverage to the main corridor. Two Powersoft Digam LD3004 four-channel amplifiers supply the power.
According to appointed system integrator Bina Teguh, “The owners of the Opera were very unhappy with the previous system installed so I was specifically asked to install an EAW system.”
“Time was of the essence, and when I contacted ProAktiv Systems, a proposal with a complete solution was submitted within 24 hours. I was very impressed.”
“In fact, the entire system was ordered and installed within two weeks of the order being made. Given the high ceilings and many varied zones, this was a hard application to get right, as we weren’t working with a blank canvas here.”
“However, the support and responsiveness that I experienced from ProAktiv Systems was a novel experience for me, having worked many years in the installation business and dealing with so many different suppliers and distributors.”
Kramer Electronics Names Kent Cawthorne As National Sales Manager
An accomplished employee, Cawthorne was promoted to the post which oversees all Kramer and Sierra Video Brand Products
Kramer Electronics USA, Inc., has announced that it has promoted Kent Cawthorne to the new position of National Sales Manager for all Kramer and Sierra Video brand products.
Cawthorne, who for the last 2 years has been Kramer’s Director of Sales for Sierra Video, a wholly owned subsidiary of Kramer Electronics USA, has an extensive background in A/V and broadcast sales.
Cawthorne began his career as a sales manager at the dealer level, focused on developing customer relationships for broadcast, production, corporate and educational customers.
Subsequently he moved to the manufacturing side of the industry and has held various management positions with such companies as Sony, Pacifiq Technologies, Panasonic and Grass Valley.
Prior to joining Kramer, Kent served as VP of Channel Sales for QuStream where he developed a variety of very successful sales strategies and initiatives.
“Kent has done a terrific job over the past two years by not only developing a dealer channel for the Sierra Video brand of routers, but also raising sales and brand awareness in several niche markets for Sierra’s new HD, SD, 3G and wideband routing switchers,” said Dave Bright, President of Kramer USA.
“In fact, under Kent’s guidance, Sierra Video’s dealer sales increased by more than 30% in 2009, which we all know was a very tough year.”
Bright further added that “Kent is a tireless salesman and manager. He has done such a tremendous job with the Sierra Video brand in just two years; it is a natural progression for him to take over sales of all products under both brands.”
“I have every faith that Kent will embrace his new and well-deserved position, and will continue the trend of double digit growth in US sales for Kramer.”
Midas XL8 Consoles For Pope Benedict XVI’s Evening Vigil & Dual BBC Events
Two XL8s were deployed at each of three events, with one console dedicated to the orchestral performances and the other for all other acts.
Midas XL8 digital consoles played a pivotal role at three of the UK’s biggest outdoor events of the year as London’s Hyde Park hosted the BBC’s Proms In The Park, BBC Radio 2’s Elvis Forever tribute and Pope Benedict XVI’s Evening Vigil.
The consoles were rented from Britannia Row by London-based System Sound, which has been providing the PA and broadcast splits for The Last Night Of The Proms and other Hyde Park events for 15 years. “This year we had the best sounding system we have ever had,” says System Sound’s Simon Biddulph. “The XL8s contributed to that in a huge way.”
Hundreds of musicians in orchestras and bands, plus singers and speakers, participated across the three events before a combined live audience of over 140,000, with millions more tuning in to the events on radio and television.
Biddulph says, “The dynamics of orchestral pieces means that any flaws are heard, while the range of musical genres in any one night means that speed of operation for the engineers is crucial. The XL8s delivered in both cases.”
In one of the most eclectic assemblages in live performance history, the XL8s processed everything from the spoken prayers of the Pope to the opera-trained vocals of Dame Kiri Te Kanawa, from the BBC Concert Orchestra to 14 year old Britain’s Got Talent star Liam McNally, from celebrated guitarist Brian May to the legendary Tom Jones singing Elvis classics.
Two XL8s were deployed at each of the three events, with one console dedicated to the orchestral performances and the other for all other acts. For the Pope’s Vigil the second console looked after everything from the three priests to the Pope’s own microphones.
Engineers were quick to praise the XL8’s intuitive user-friendly design and singled out the POP(ulation) groups as one of the console’s stand out features, enabling them to work quickly and confidently under pressure.
“The XL8 was quick and easy to use,” says Richard Sharratt, who mixed Proms In The Park. “The great thing about the POP groups is that it doesn’t really matter where an input is, – it’s always instantly accessible.”
Chris Coxhead, who mixed the bands for Elvis Forever and vocals for the Pope’s Vigil expands: “The POP groups and VCA assignment allows you to get to channels very quickly and the ability to ‘park’ stuff in area B means those very important money channels are always available.”
Meanwhile Colin Pink, who mixed the orchestras for Elvis Forever and the Pope’s Vigil, praised the noise floor and seemingly infinite headroom available when mixing more than 90 channels of instruments.
Coxhead particularly appreciated the onboard FX and dynamics. “I inserted the onboard multiband compressor across the subgroups which had the lectern, altar and Pope’s mics so I could reduce the proximity effect if anyone got too close. It worked very well,” he says, not without some relief. “After all,” he explains, “it’s not every day you have ‘Pope’ on your inputs.”
Universal Audio Releases The EP-34 Tape Echo Plug-In For UAD-2 Platform
The warm, adjustable, tape echo emulation is inspired by two classic Echoplex units.
Universal Audio has announced the release of the EP-34 Tape Echo plug-in for the UAD-2.
Universal Audio’s EP-34 Tape Echo plug-in gives guitar players and mix engineers the rich, warm tape delay effects of vintage Echoplex units, now on the UAD-2 Powered Plug-Ins platform for Mac and PC.
Unlike other Echoplex emulations, the EP-34 is the first plug-in that targets specific behaviors of both the EP-3 and EP -4 Echoplexes.
The DSP experts at UA ensured that the EP-34 plug-in is verifiably the most accurate model for those who want the distinct, chaotic Echoplex sound, “warts and all.”
“The rich tape echo sounds and tactile controls of the original EP-3 and EP-4 hardware units have been cherished by musicians and engineers alike for decades,” said Matt Ward, President of Universal Audio.
“We’re elated to be the first to provide the dead-on sonic qualities and controls from both of these classic units as a software emulation.”
Available for purchase via UA’s Online Store, the EP-34 Tape Echo Powered Plug-In features include:
• Highly desirable sound of original EP-3 and EP-4 Echoplex tape delay units, now available as a UAD-2 plug-in for Mac or PC
• Tape echo effects virtually indistinguishable from those used on countless classic recordings of the past six decades
• Unique movable record head design creates warm, rich sound unlike any other delay unit
The EP-34 Tape Echo plug-in is available as part of the new UAD Software v5.7.0 release. In addition to the EP-34 Tape Echo, v5.7.0 includes the now officially licensed dbx 160 Compressor/Limiter plug-in, as well as additional performance enhancements for all UAD-2 users on the Windows platform.
More information on the EP-34 Tape Echo Powered Plug-Ins for UAD-2 can be found on the Universal Audio website.
The second video in a multi-part series from our resident church blogger on some of the basics of church audio.
From our resident church audio blogger Gary Zandstra comes the second video in a multi-part series on the basics of church audio, which is equally useful to novices in any field.
For more helpful hints from Gary, make sure to check out his Church Sound Blog.
Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 25 years.
The C6 features four crossover and two floating bands plus a sidechain feature, making it an ideal tool for studio, live sound and post production.
Waves Audio has announced the introduction of the C6 Multiband Compressor.
For years, the Waves C4 has been used engineers the world over and the C6 builds upon its predecessor’s functionality, taking it to the next level.
To create the C6, Waves took all the functions of the C4, and then added two additional floating bands plus a sidechain feature, for one-stop vocal and instrument shaping.
The C6 lets users zero in on problem frequencies with surgical precision. With four crossover bands plus two additional floating bands and a flexible sidechain feature, the C6 is an essential solution for vocal and instrument shaping, offering de-essing, de-popping, feedback management and more, giving users all the multiband compression and dynamic equalization they need to control, tame and shape their sound.
The C6 is hugely versatile and will find use among studio, live sound and post production engineers.
For mixing and mastering in the studio, the C6 gives users the power to shape any sound, with creative compression, dynamic EQ, flexible sidechaining and more. For live sound, the C6 provides total control over dynamics and EQ, from feedback management to in-depth vocal shaping.
For auto-ducking of music and ambience under speech in a post production setting, the C6 sidechain lets users carve out just the vocal frequency range, instead of attenuating the entire signal.
Features include:
• Internal/external sidechain per band
• Individual band Listen mode
• Four crossover bands plus two floating bands
• Dynamic EQ, compression and expansion
• Double precision bit resolution processing
• ARC™ Auto Release Control
• Up to 24-bit, 192kHz resolution
• Mono and Stereo components
• Supports TDM, RTAS, Audio Suite, VST, AU
• PC- and Mac®-compatible
“The ability to have six bands of fully sweepable dynamic EQ on the mix bus makes the C6 an all-important tool for live sound engineers”, said FOH Engineer and Waves Live Division Product Specialist Ken “Pooch” Van Druten.
“By itself, it does the work of two or three different plugs that I used to use.” DJ/Electronic Musician Luke Slater added, “I love the C6. Very musical, very intuitive!”
Waves C6 Multiband Compressor is available separately and as part of the Waves Mercury bundle. Mercury V7 owners covered by Waves Update Plan receive the C6 at no additional charge.
You see, I’m just starting out in the A/V field and I really want to make sure I’m equipping myself with the best set of knowledge from the get-go.
So, I have a question for you, which may be simple (and a little open-ended) however I could really use an explanation.
What is impedance balancing?
A: First of all, what a great sentiment! I’m glad to hear you’re taking such a proactive approach to your professional growth.
One reason you may be a bit confused with the term impedance balancing is that it has been known to go by more than one name. It is also often called pseudo-balancing, quasi-balancing, resistor balancing, ground compensated balancing, or any of a dozen other similar names.
There are some differences between these, but basically it’s a method of creating an output that will function similar to a balanced output without having to employ all of the electronics normally required in a fully (differentially) balanced system.
The benefit is reduced cost and sometimes increased overall flexibility without losing much performance in many situations.
Here’s what it is and how it works: In a truly balanced output there are two conductors and a ground wire. The two conductors each have the output signal on them, but at opposite polarities.
Having this signal on both wires is beneficial for several reasons, but one of the biggest benefits is just in having the two wires, even if signal is only on one of them. A balanced input is able to look at the hot and cold wire’s signal and compare the difference between them.
Any differences get amplified; any signal that’s the same between the two gets cancelled. That means noise and hum picked up in the wires along the way will get cancelled, and any signal - even if it’s only on one wire - will get amplified. It’s a great system, but comes at a cost in terms of the components required.
If you’ve understood everything so far you can now see that you can drive an unbalanced output into a balanced input and still get much of the benefit of the CMRR (Common Mode Rejection Ratio) of that input.
In such a configuration it is common to tie the ‘cold’ wire going to the balanced input directly to the ground at the source end of the wire.
Any noise picked up along the wire can get cancelled at the other end.
However, because the cold wire is tied to ground the noise picked up is more or less killed right there, or ‘shunted’ to ground as they say.
Consequently most of it doesn’t show up at the balanced input, which never gives that input device the opportunity to use it to cancel the same noise that was pickup up on the ‘hot’ signal wire.
Further it opens up the possibility for noises on the ground line to get into your audio through the negative side of that balanced input.
In a ground compensated system that cold wire is not tied directly to ground. Instead a resistor is placed in between the cold and ground so that noises picked up in the wire are not shunted off to ground. Now they appear at the other end just as they would if the line were a differentially balanced line.
They get cancelled and the noise mostly goes away. The effectiveness of this scheme is largely dependent upon the CMRR ratio of the input device and the accuracy of the resistance in simulating a source impedance that’s the same as the hot wire.
This is not a truly balanced system, but in terms of noise cancellation behaves similarly.
If you are working with a device that employs impedance balancing you should connect it to other balanced devices just as if it’s a normal balanced output with hot, cold, and ground leads.
If you are connecting its output to an unbalanced device you can use a standard unbalanced cable (which actually makes it easier than connecting a balanced output to an unbalanced input [what to do with that third conductor]).
Some equipment can use its ‘cold’ output terminal as what is known as a ‘ground sense’ line.
Thus even when driving an unbalanced input (where you’d have the cold terminal tied to ground) this sense line is able to add any noise or hums from the ground back into the hot signal where they would get canceled at the other end.
They get canceled there because the same signal would be appearing at the ground of that device (since it came from the ground wire in the first place). In unbalanced inputs, ground serves as a reference against which the signal on the hot lead is taken.
If the hot and ground have the same waveform on them then nothing appears at the device to amplify, and again the noise ends up getting canceled. This isn’t as effective as a balanced input, but the performance is certainly better than a typical unbalanced line.
As always, we welcome input from the PSW community and would love your thoughts on the topic at hand. Feel free to let us know in the comments below.
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