Church Audio Blog

Show file Templates

noreply@blogger.com (Anonymous) — Wed, 08 Oct 2014 00:00:00 +0000

I'm trying to create a new show template for volunteers to use as this is something that came up in a recent breakfast meeting with the volunteers. Up to this point for the past two years we have been using a basic template that slowly evolves each week by recalling whole channel presets for instrumentalists and vocalists. There are pros and cons on both sides of the table.

On the one hand changes to the template are captured each week and you start from last week and don't have to reinvent the wheel by starting from a new template by dropping in presets every week. There is weekly momentum. On the other hand (and we have experienced this) if a setting is screwed up, it can sometimes stick for a week or two before it is properly sorted out.

Below are some of my initial notes and thoughts on what I'm looking to do to modify our weekly running show file into a "basic template" that we can run from every week. Any additional thoughts please let me know in the comment section.

New Show Template: the advantage is that it would ensure (hopefully) that everyone is using the same template for Sunday morning. The disadvantage is that the FOH is required to invest additional time setting up a show template rather than making small changes to the previous week’s template. The goal is to provide a framework to get to a final mix faster and produce a better sound through consistent workflow. Mixing is not a cookie-cutter process.

It should include updated patching for PT and Stage for Virtual Soundcheck

All channels should be mapped, patched and labeled.

Can the template be write-protected??

How long should it take someone to set up week-to-week changes in the software? 5-10 mins?

What should be preset versus manually set in the template?

- Recalling whole channel presets should be manual

- combining/splitting mono strips to/from stereo should be manual

- building Custom VCA (optional) should be manual

- Basic snapshot structure should be preset

Squiggly Lines

noreply@blogger.com (Anonymous) — Sat, 01 Mar 2014 23:28:00 +0000

A few weeks ago I got a special project from our pastor to create a map of the seating chart that accurately shows the loudness in the seating area of our main auditorium that shows the overall loudness for each seat in a way that the ushers could use to show people to seats that are louder than others. Since we make all of our measurements in dB-A weighting slow it made sense to use make an A-weighted measurement of each seat within the seating area. Looking at the graph below (that shows the results) shows both some interesting patterns. I set the PA at 80dB at FOH and then proceeded to walk the room. The loudest seat is Block 1, Row 5, Seat 6 at 85dB. The quietest seat is in Block 6 just infronto f the tech booth at 79.6dB.
Seeing this graph got me to thinking what the frequency response at FOH was. I had never measured it before. When I did. I was very surprised!! The graph to the left shows a spectral decay of the original FFT and the it's repsonse decaying over time in 20 millisecond increments. I'm still learning about acoustics, which is like drinking from a fire hose.

Looking at this graph tells me a few things. First the Subs are about 10dB louder than the main PA. Second the frequency response at FOH is not linear, but that's not something I want to tackle. I would like to do some more measurements in some spots of the room but I may want to see if we can have an acoustics consultant come to re-tune the room. Or perhaps it is a result of the PA needing a tune-up and putting some replacement drivers in the cabinets. Speakers are moving parts after all!

Jagged Lines

noreply@blogger.com (Anonymous) — Fri, 07 Feb 2014 17:05:00 +0000

Having a few Amazon gift cards from Christmas and a spare laptop (hey, I'm a tech guy I've got a a lot in my house) I decided I wanted to start understanding more about Acoustics on a practical level. Having bought and read (most of what I can comprehend) of the books, Master Handbook of Acoustics, by F. Alton Everest, Sound System Engineering, by Don Davis, and Sound System Design, by Bob McCarthy I thought it was time to put the theory to practice. There is about nine or ten posts I could do on acoustics but to keep things simple I want this first post to be practical because most of us in churches have a 31 band EQ or some flavor of DSP strapped across the speakers.

So I bought a USB interface and a measurement mic and I needed software. I just don't have the money to pay Rational Acoustics for Smaart 7. Maybe someday, but not now. I downloaded REW, which offers probably 75% of what Smaart can do for a lower investment. You still have to calibrate your measurement mic, and create a calibration curve of the noise floor of your USB interface, but the functionality is there without the cost.

I started with the stage monitors because no one is every happy with the way they sound and using music to EQ is never objective and I'm not going to touch our main PA without bringing in a trained and experienced acoustician. So Here's what I have learned so far - when it comes to processing, less is more. Looking at the graph for the first time and seeing all the sharp jagged response that looks like the Himalayas you can be tempted to go wild with an EQ, but I have the resolution set at its highest (1/48th) octave smoothing view. If I were to change that to 1 octave smoothing it would look like a flat frequency response that rolls off around 80Hz. This is the picture of the response AFTER applying EQ.

You have to remember that if you start with a bad sounding loudspeaker you aren't going to be able to do much, and there is a temptation to go crazy with EQ. The second is the EQ processing that I applied to the EAW SM129 wedge, the horizontal hash marks on the Ashly Protea denote 5dB increments. So as you can see, the most damage I am doing at one particular frequency is -3dB at 5kHz.

I know that we have speakers in our youth room that have torn HF drivers (i.e. tons of distortion) so I can't wait to see what the amplitude and phase relationship look like when I measure it. There is so much I have yet to try I have not even scratched the surface.

The RF Stuff

noreply@blogger.com (Anonymous) — Sun, 03 Nov 2013 21:13:00 +0000

Our church is looking for wireless microphones. We have been working with the tried and true old Shure ULX for about eleven years now and after a few weeks of some interesting mishaps and a long time of it being less than reliable such as our pastor sitting on the plastic transmitter pack and crushing it I think it’s time to start the hunt for a new system. We bought a new pair of lapel belt packs but we have certainly got our money’s value out of our current system.

What I want in a system is something reliable, scalable, affordable, and I want it to fit in a 1U rack space… if possible. There’s a lot of buzz around ULX-D, but looking around I saw a wireless system from SACOM that looked interesting. I know for a lot of churches the bottom line is the bottom line. Cost is a huge issue. If you are going to invest thousands of dollars in a new wireless system you need to demonstrate its value-add. We are no different.

Here is what I like from SACOM:

1. Scalability: you can short-load their 8-channel receiver with blanks and start with one or two channels and scale your system over time by adding more receiver modules and transmitters later. You can also cascade up to eight 1U receivers on two antennas giving you 64 channels with the Ethernet in and out.

2. Mission Critical: This is a phrase people throw around, but hearing from John at SACOM that this unit not only uses 256-bit AES encryption but conforms to FIPS 197 and is in use in several US Government buildings because it meets specific criterion for auto-failover, programmable redundancy and web or test messaging for status alerts. That’s impressive. You can find most of these features in Shure’s Axient line, which has a MSRP of, wait for it… $9,299 for the AXT400.

Shure		ULXD4Q
MSRP LIST PRICE		$ 6,269.00
Per Channel Cost		$ 1,567.25

SACOM		DS88-E-N-S-N-U-M915
MSRP LIST PRICE		$ 5,199.00
Per Channel Cost		$ 649.87

3. COST: What I’m listing here is MSRP PRICE, not street price as that will vary over time and from one dealer to the next. I’m also not listing any accessories like handheld microphones, lavaliere belt packs or antennas as your system configuration will look different than mine but take a look. I think the numbers speak for themselves.

Keep in mind, the Shure wireless is four channels, SACOM’s is eight channels. So when you look at the per-channel cost between Shure and SACOM there is a big difference. So if you short load a system and remove receiver modules to make it a four channel system that would be comparable to ULXD, every time you remove a receiver module it reduces the system cost.

There’s a catch, isn’t there?? Well, sadly yes. Unlike Shure, SACOM doesn’t sell big box business to point and click online AV dealers like B&H. That’s not their business model. Send them an email asking who your nearest SACOM dealers are to get in touch with for designing a wireless system that works for you.

NAMM 2013 Recap: More Fun Stuff

noreply@blogger.com (Anonymous) — Mon, 04 Feb 2013 21:24:00 +0000

I'll admit that not all these are my pictures however there was simply too much for me to see. I only made it on the ground floor of the show and part of the basement. There was still the second floor and third floor that had lots of booths I didn't see.

First I thought this was the Holophone Big O but that is in the left of the picture as the long instrument SM57 looking microphone. This large 19th century diving helmet looking microphone is something brand new to NAMM and comes in either cardioid or omni, and is their answer to making microphones look cooler on the stage. Their reasoning is that the SM58 looks too boring. It looked very unique, and I remember the price point on this was right around $1000, but then again everything Holophone makes is up there in price.

These Renkus Heinz IC2 arrayable speaker coffee tables grabbed my eye and I thought that no green room would be complete without something like this. Heck, I'm thinking this might go good in my home office. This also puts old broken gear to some good use. We have a rack of old analogue compressors and EQ's that I can't sell for how old they are just sitting in the back. Perhaps turning the rack on it's back and putting a piece of glass over it like this would be a creative idea for your musicians green room.

This idea I found very interesting, using rope-light inside the kick drum to add a lighting effect to your drummer and spice up your stage design. Maybe if you have a small cheap LED fixture, you could use it to match the color wash on your stage and run the power and DMX out the kick hole or Tom stand mout hole if the kick drum has one that is not being used.

Guitar Boost Compression

noreply@blogger.com (Anonymous) — Sun, 03 Feb 2013 16:54:00 +0000

One thing I tried not long ago was to smash electric guitars during boosted guitar solos. I can’t remember where I heard or read about this but it made sense. People who know me or have followed this blog have learned that I’m not a fan of the hyper-compression mentality that “every channel has a compressor, so I need to use them.” Or the, “if at first you don’t succeed, compress!” No. It has to work in the context of the song and the mix.

Why compress electric guitars? For a lot of electric guitars (especially tube guitar amps) the tone comes from the sound of the tubes starting to break up / saturate / distort, which means there’s less dynamic range. In the past I asked our guitar players to adjust their boost levels up or down (usually up) for them to control their boost levels during guitar solos. The reason being is that it was one less fader to ride. And it worked very well, and still does. I just wanted to try something new.

Our guitar players typically use a distortion pedal (the Suhr Koko Boost, Suhr Riot, or Fulltone OCD) and run their amps clean so that they still have headroom to boost. Then they will boost into tube break up, that way they gain extra tone when saturating the tube.

I’ve found that many times guitar boosts bring out the upper midrange frequencies that typically cut through the mix tonally. Our guitar players have somewhat bright Marshall-ish sounding tone. So on a bright sounding distorted guitar this can sometimes sound a bit unpleasant as it over accentuates the critical band between 1kHz and 4kHz where our ears are most sensitive. Squishing the boosts to get a fatter tone while still getting more dB’s coming from the boost helps me counteract the piercing quality that some guitars can have. Again, this is just a preference thing; it may not work for your guitar players.

I first tried the onboard compressor but couldn’t seem to get it to work the way I wanted to and then moved to the TL Aggro plug-in. With a bit of tweaking I found a setting that worked good on the guitar during boosts to make the tone sound larger without a lot of extra volume. For our guitar players I found that a ratio between 3:1 and 5:1 worked. I still need to push the fader during guitar solos, but the compressed tone that is more in-your-face from being compressed does a bit to help. The way I have the compressor set up is during normal playing I’ll see the gain reduction meter gently tickling the signal around -1dB or -2dB. During guitar solos when the guitarist is boosting it will be around -3dB to -7dB.

NAMM 2013 Recap: More iPad Geekery

noreply@blogger.com (Anonymous) — Fri, 25 Jan 2013 18:08:00 +0000

Yesterday I spent the day at the NAMM Show in Anaheim. To sum up the NAMM show, I believe this is the year of the iPad. It seemed almost every product, solution had a digital interface or app that goes on your iPad. I think the audio and music manufacturers are starting to make things more open platform, realizing they can cut costs of developing hardware by having a software apps available for your iOS device. Here is a great example of that:

The MicW i436

Following up from my post on iPad dB Meters, the one thing I talk to people about is how using your iPad or iPhone as a dB meter can be dangerous. Why? Well, it's not a calibrated reference microphone, it's a pinhole mic built into your telephone it isn't a calibrated dB meter. The designers of the software audio analysis tools did a great job writing algorithms to account for the nonlinearities in the iOS device's frequency response. But it still leaves the problem of a built-to-budget microphone.

I took a look at the specs on the MicW website and they claim that the i436, "is a calibrated measurement microphone complying with the IEC 61672 Class 2 sound level meter standard." I am still hesitant about using your iPhone or iPad running the Studio Six Digital or Faber Acoustical analysis apps for true analysis of SPL or FFT analysis. I'm interested in this because I use the FFT plugin in the Audio Tools app for a visual reference of what my ears are telling me. It's also considerably cheaper than the Studio Six Digital iTest Mic. The mic comes in two forms. Mic only has a street price around $120. The kit comes with a case, windscreen, 6' cable extension, mic clip, and keychain case that also doubles as a mic stand. The kit has a street price around $160.

Here Dan from the Full Scale AV booth was demonstrating the iMic line. They also have the i456 cardioid, i266 high sensitivity cardioid, i825 omni lavalier, i855 cardioid lavilier, and the iShotgun which can mounted on a DSLR to improve audio for video capture. I think this might make for a good competition to the portable digital recorders on the market for a little less money and it's still cheaper than many recorders.

There is way too much to cover about NAMM in this one post, I'll try and follow up with another later.

Moving On

noreply@blogger.com (Anonymous) — Mon, 26 Nov 2012 19:15:00 +0000

It’s often said that hindsight is 20:20. It’s pretty amazing to see how God has worked in my life to this point, how he introduced me to my wife; how I got to the place I am now, the ways he has protected me over the years. The divine fingerprints are pretty clear to me the calling God placed on me to be where I am and the job I do. If only looking forward were as easy as looking back.

On September 25th I was informed that on January 1, 2013 my position will be dissolved. I’ll be leaving my position at Sunridge and I am trusting that God will put me where he wants me. I’ve seen many situations in life that don’t make sense, and then months or years later find out there was a reason God made that move.

On one hand it’s the one thing all of us as paid church techs fear most. You try and work yourself out of a job (building into other people and delegating not just a task but ownership) and the church could decide that your team can function without you. On the other hand I believe so much in volunteers and empower them to own the ministry I have been throttling back the amount I am doing over the past few weeks and over the next month and barring a few small hiccups, the team is owning the tech ministry. And it’s really cool. Lately I’ve been feeling like I’m not a good leader of volunteers, but seeing the team keep pace as I take a step back makes me feel proud that I’m not a failure; I have done a good job.

Right now I’m looking and praying about what is next for me. It will also be difficult for me as I've been at Sunridge for 22 years, and I love this church. We need your prayers as we move through this change. As you can imagine, it's still a very tough economy and I’m struggling to find full-time opportunities. If you know of a secular organization or church looking for a full time audio guy or technical person please let me know.

Communication & Documentation

noreply@blogger.com (Anonymous) — Wed, 31 Oct 2012 17:33:00 +0000

One of the challenges we have had to deal with in the past is a lack of understanding on how the system in our church has been put together because no system maps, as-build drawings or schematics were ever delivered when we moved into our facility. As a part of my job to document and pass on information my boss has asked me to document and map out all of the procedures and systems for audio, video, lighting, ProPresenter, and our Youth Room system. Since we just did a major overhaul of the audio system in our main Auditorium and now that I know where everything is, it seemed best to start there.

I've included linked files from Dropbox of the finished system maps of how the SC48 is patched and the system map of the outputs to the monitors, speakers, and external outputs (hallway, green room, cry room, video… etc.)
Audio System Map.pdf
SC48 I/O Patch Sheet.xlsx

The System Map was simply too involved to create it on one page. I broke it into three pages. Page 1 is all of the monitor outputs for the band with the wedge, IEM, and wired In Ears. We use our the HearBack system for Keys and Drums and are planning to integrate the Guitars in the next few months, so it's shown here as a future item. The same is true of the Sennheiser G3 IEMs for IEM 7&8. We have a spending cap on how much we can spend in the first quarter of our budget year so the last set of IEMs will go in after Jan 1. Page two of the system is all the external outputs. It wasn't absolutely necessary to create this page but I thought it made sense as it helps to look at this and the Patch Sheet together to gain clarity on how things are laid out. Page 3 was the most important for me as the Amp racks in the electrical room were labeled by the integrator using the AES standard for labeling in audio system integration. However I'm the only person who understands that "C-LT-1" is the center cluster cabinet 1 and that "DS-1" is delay-fill speaker 1.

Looking at the Patch Sheet we did a lot of standardizing. We realized that we were reinventing the wheel (and the mix) every week and by keeping the staple inputs coming down the same lines and standardizing where other inputs would go makes it more volunteer friendly as last week I had one of my volunteers come in and do production for a memorial service and with these documents, he came in and rocked it only having used the SC48 a couple of times. All of this documentation has been laminated and punched and is living in a 3-ring binder that lives in the tech booth. I know these system maps are not exhaustive or look like a professional did them, but they are designed to be volunteer friendly. I used the free software AV Snap to create these system maps. What I really love about AV Snap is that you can copy and paste photos of the products you actually use in your room into the program to make it easier to read. I'm a very visual person so seeing the Ashly Protea's in line between the console and amps helps me understand where they are in the monitor system. You can also standardize templates and pages to make adding elements to your system easy.

Hello Digi! Sorry, I mean Avid!

noreply@blogger.com (Anonymous) — Fri, 17 Aug 2012 18:51:00 +0000

This past week we made a huge jump in our audio world moving from the Allen&Heath ML5000 to the Avid SC48. If you follow me on Twitter you probably saw some of the photos I posted. And if you are still following me on Twitter I need to apologize if you got spam from me, my Twitter account was hacked. It’s been a loooong week. After a quick sum I counted that I put in almost 60 hours between Sunday and Thursday soldering wiring, testing and setting levels for all the inputs and outputs to our various destinations… etc. I'm glad that I wasn't mixing for rehearsal last night because it felt like I had a bit of an ear infection. Most likely from the exhaustion and lack of sleep.

About a month ago we bought the SC48, however we decided to schedule the install for this past Sunday afternoon. This gave us time to label every cable going in and out of the console so that everyone knew what was staying and what had to go. Then, after service, at 1PM we came back and started pulling the old console. This meant all our analogue outboard of compressors, Effects units, graphic EQ’s had to go. That, along with all the wiring and channel insert sub-snakes that connected them to the console. In short, it was a giant spaghetti mess. After we cleared that it was down to making sure (one last time) that there weren’t any cables that were unlabeled and mysterious and we pulled all the outputs, and channel inputs.

Our old FOH console with outboard.

We then realized that an old analogue console is about as heavy as a baby rhinoceros. It took all 6 of us to slide the console off the back and on to a rolling pallet. From there we did a lot more de-tangling of cables and dusting before 3 guys landed the SC48. It’s a much lighter console. I also want to give a huge thank-you shout out to my audio guys and some of the band members who helped with the install and especially Eric Kibbe the Audio Manager from Saddleback who came and helped us with the set up and building the initial show files and templates for the console and Pro Tools. This week would have been a giant mess without his help.

The one little snag we found was that when we started up the console Mix Engine 2 had failed so we called the Avid support line and they over-nighted the card (which is essentially an Avid HDx card but with ribbon cable ports instead of the HDx ports). The Avid technician came and while the Avid man wasn’t looking I snapped a quick photo of what the console looks like with “the hood open.”

Avid did a great job supporting us. Within 24 hours of calling we had the part in hand, the technician on-site replacing the piece, and the console working again. Both Eric and the Avid technician said that having a part defective on a new-in-box product is incredibly rare. The learning curve is that if you get a new console, as soon as you power it up, run the Digitest on the System page to see if everything is working properly. If something is not working correctly, call the support number.

The last few things that need to be done is to order an uninterruptible power supply (UPS) to put the console on. Here in SoCal during the summer months power outs and brown outs are not uncommon. And if there is a power outage, having a UPS gives us enough time if there is a power out or brown out to save our work and shut the console down.

The new FOH with cleaned up racks and the SC48.

I’m also waiting for more RAM and a larger hard drive to arrive that I ordered for doing virtual sound checks. It worked last night but was a little squirrelly. The first time we fired up the Mac Mini and launched ProTools it told me I needed more than 2GB of RAM. Hmmm.

I also need to order a 15’ AES cable to use that will go from the console into our TC Finalizer for the podcast and video mix. Right now it’s running into the analogue inputs, but using the AES out saves us 2 outputs for future use.

Lastly, I’m looking for an audio DA (distribution amplifier) that has volume control on each output. Right now we are using several Matrix outputs each to feed to the green room, cry rooms, hallways, and Youth room. However those can be just one mono matrix with independent volume controls for each output if we get a DA. Drawmer makes a nice DA called the DA6, however it’s street price is around $800. That's a lot more than I want to pay. I’m thinking the sub $300 range. If you can think of something that works, add your thoughts to the comment section.

Other Than God, Whom Do You Serve?

noreply@blogger.com (Anonymous) — Tue, 10 Jul 2012 00:33:00 +0000

About a week ago I sent an email to some people on my tech team asking for some feedback (the good kind) about what I do well and what I can improve upon. It's one thing to have the leadership tell you to improve on areas they see are important but in a volunteer organization I think you should also know how to serve and lead your volunteers better. And that occasionally means asking. I simply asked what they think I'm doing well and what they think I can improve upon.

To be honest, I was initially hesitant because when I asked two of my volunteers almost a year ago, one came to me with a laundry list of issues with my leadership style. Some of them were a little nit-picky, but some were revealing and true and his observations have helped me grow in leadership and serving my volunteers needs. So here we go again. And one of my volunteers that responded had some amazing observations on what I do well and can improve upon. And, as the post title indicates, the question is, "Other Than God, Whom Do You Serve?" That question really made me pause for a while and think, whom do I really serve? Is it my boss and the pastor? Volunteers? Whom do you serve?

Chaos and Order

noreply@blogger.com (Anonymous) — Sun, 24 Jun 2012 22:49:00 +0000

World History in High School was a bore. But I do remember one thing Mr. Whatsisface taught me, "one minute of organization saves two minutes of time. There are so many things that I run into as an audio guy, not just at the church but as a professional as well that my bag I travel with needs to be somewhat organized.

What I have in my bag I travel around with is a few things. First at the back is my Dell laptop, which I use for doing anything that needs a Windows OS to work with: QSC Signal Manager, the Avid D-Show software… you get the idea. Next is my iPad that has an Etch a Sketch case. Then you notice I have several leather bags. One of them is my headphones, the reset are microphone bags. Since all our microphones are in a drawer I found that I had a few collecting dust. And rather than just shoving stuff in the bottom of my bagIn these bags I have different bags that contain several different things. One bag has a few cables to charge my iPad and laptop. Another bag has a mouse and some common cable adaptors I usually need. The third bag has Sharpies pens, USB sticks and a few SD cards and a card reader I use as a featherweight external hard drive when I need it. How do you keep yourself organized? What creative ways do you maintain order in your ministry?

The 10V battery in your 9V Wireless

noreply@blogger.com (Anonymous) — Fri, 15 Jun 2012 16:06:00 +0000

So if you missed my former post on 10V batteries not working in Shure ULX handheld microphones you might want to go check it out. I sent in some wireless microphones to Shure for repair and when I got one back I thought about sending it in again because when I put our rechargable batteries in the mic it freaked out making a wobb wobb wobb sound.

Last week I determined that our 9V batteries actually arent. They're 10V batteries, and when I tossed in a duracel I found lying around that still registered 9 volts and it worked I decided this week to try and drain one of the batteries to (if you look at the meter and squint) it reads 9 volts.

At that point I put the battery in the microphone turned it on and it worked perfectly. This confirms what I've been suspecting and others have been telling me that the operating line voltage of the ULX microphones is a bit sensitive to voltages above a 9V battery. So for that specific handheld microphone I'll need to remember to either keep one alkaline battery handy for using it or drain a rechargable a little bit so that the microphone doesn't freak out.

Do Wireless Battery Voltages Matter?

noreply@blogger.com (Anonymous) — Fri, 08 Jun 2012 15:45:00 +0000

Has anyone ever dealt with this?

I had a few wireless mics that were intermittent in signal so I sent them in to be repaired. I get one of my ULX handheld microphones back and I put one of our rechargable batteries in the mic and it freaks out a bit. It either passes a "wobb wobb wobb" sound or nothing. I put in an alkaline battery and try it again it works perfectly. So I tested the two batteries. The Duracel measured 8.9 volts. The Ansmann rechargeable measured 10 volts. That's something a bit high for a 9V battery. But does that matter?

My next experiment will be to put the 9V rechargeable in another device until it reaches the same 8.9 volts and then try it in the microphone again. Wireless mics will continue to work as a battery is dying. That said, I suspect that the internal circuit board in the hand held microphone has a small tolerance for more than 9 volts of power, which is why it is freaking out. Does anyone have an explanation on what may be happening?

The Art of Gimmickry

noreply@blogger.com (Anonymous) — Wed, 30 May 2012 15:37:00 +0000

So much of the entertainment industry is based upon using the latest trick or technology to grab and hold our attention. It makes the assumption that we have to make the “thing” look and feel interesting for it to actually be.

It seems like in the late 1990s to early 2000’s I remember magazine covers used neon bright high-contrast colors and fonts to grab your eyes hoping you’ll pick up the magazine, flip through it and take it to the checkout. Did it work? Yeah. How about now? No, people are catching on to the gimmick. Sadly though, Magazine companies are on to the next gimmick.

Church is not like going to a movie theater or a concert where you pay money for an experience. People are coming because they want to. And it's free. And that’s not to be confused with the thought that people just want free production on Sunday morning. Are we using or searching for the latest gimmick to hold people’s attention in hopes that they’ll come back next week? Or that maybe they’ll invite their friends? Ok, but as the question, why? Why do we want people to come back next week? What do we hope to accomplish during the one-hour service in the lives of those people? What won't happen if people don't show up next week?

Church has to be about life change, moving from death to life, it’s about grace. If it’s about anything else, it needs to go back to the drawing board. What are your thoughts?

A Tale of Two Digi's… The Avid SC48

noreply@blogger.com (Anonymous) — Thu, 29 Mar 2012 18:47:00 +0000

Doing the review I spent an afternoon at Saddleback Church. There, I got a great education on what the SC48 does, how they integrated the VENUE system in different ways throughout their various worship spaces and campuses and why moving to Avid made sense for them for what their direction is as a church. Though we didn’t get to spend much time passing audio through the console the time was beneficial and gave me ideas on possible ways to integrate it into our system.

Last week, I set up some time with a Tech Director from a church a few miles from me. They recently installed an SC48 and I spent a few hours in virtual sound check doing some playing around, mixing. The first half of the time I was getting frustrated with the desk paging around layers until I remembered that to work faster I should actually be using the VCA Group spill features. After that I noticed I was able to build my mix a bit faster. That is until I actually starting working with the channel functions such as EQ and dynamics. I found the knobsets on the ACS to be a bit too clunky and they didn’t respond to how I wanted to touch them. The horizontally orientation of the knobsets becomes especially confusing for us analogue veterans when it comes to EQ’ing your instruments.

It wasn’t until I starting working with plug-ins that I got frustrated with the console. And that’s because I ended up spending the majority of the allotted time fiddling with plug-ins hoping I could make the guitars, snare and lead vocal better instead of finding a good workflow on the console and concentrating on the big picture of the mix. This meant that the level balance, frequency response and direction of the song suffered after almost three hours of mixing 3 songs. By that time I should have a finished product mix when it comes to live sound work.

Starting off this process of looking for digital console I was already set on the SC48 being the right choice when I had not yet taken a good hard look at what it really does, let alone how we would use it. The big question I keep asking myself is, what am I really paying for? So many consoles do the same thing: library presets, snapshot automation, GPIO control, VNC control, virtual sound check, remote stage racks. So if they all do the same thing, at the end of the day it’s all about the workflow of how it automates mundane processes to let you get to the core of mixing for your event. What are the big selling points of this? And with the Avid it’s two things: plug-in's and ProTools.

Plug-in’s

For plug-ins, if it’s a TDM plug-ins and will work on a PC computer running ProTools, you can run it on the console. And while there are a few Mac only plug-ins, the vast majority of plug-ins are platform androgynous. Talking with the Tech Director who let me mix via virtual sound check for a few hours, both he and I are a bit annoyed by the fact that if you want any plug-in you have to fork over some serious coin. He mentioned how he would love to use some of the plug-ins from the Waves Live bundle but given that the price is over $1500 his comment to me was, “who can afford this stuff?” The average price of any plug-in is at least $100, and I know that not just me and him but many church tech guys will have a difficult time justifying to their leadership several times over the years how investing several hundreds of dollars, or thousands cumulatively will give a noticeable and marked improvement of sound quality in their mix. That money could be also spent on better microphones, DI’s, saving for a new PA system, etc.

ProTools

I have to admit that I am biased in this way. Being a former studio guy, I first learned ProTools working part time at a studio in college when we were on Pro Tools version 5. I know the software, and I love it and I used it for years working in studios. The great thing about ProTools is that it’s like learning the guitar. If you want to noodle around and just get going with recording or mixing the initial learning curve is fairly easy to pick up. But if you really want to dig into the features and learn all the little ProTools shortcut keys it will take time to learn.

For the purposes of virtual sound check you won’t be doing any Pro Tools editing. So the argument that many make that you’re doing virtual sound check on “the industry standard” DAW is a moot point. The purpose of virtual sound check is to record your inputs, and then play them back through the console to refine your mix or train a new user how to operate the desk, it has nothing to do with the software. A new feature in linking the SC48 to ProTools is called VENUE link. If you open a new ProTools session for virtual sound check when you are running or setting up a show, ProTools will automatically retrieve the channels in use and the channel names and apply them to the recording tracks in ProTools. No need to sit around naming your tracks, now that is cool!

Remote Stagebox

There are actually two options on the SC48, now that Avid has released the option to use their Stage 48 remote rack to put your inputs closer to the stage. This is been something that since day one of the SC48 release Avid has taken a lot of heat for. Originally it was designed as a console that would drop in and replace your analogue console. But the fact is, every console manufacturer is integrating remote stage boxes with their consoles. The reason being is, 1) you don’t have an expensive, heavy copper snake you have to pull from the stage to Front Of House. 2) Putting the microphone preamps closer to your microphones gives you shorter cable runs and reduces any interference, impedance, or capacitance issues that could creep into your signals from long snake runs, giving you a cleaner signal.

This is particularly beneficial for new installs because you can actually save money by not having to purchase a big copper snake. But if you already have the copper pulled to FOH, the SC48 without the Stage 48 probably makes more sense. The other benefit of having the Stage 48 is that it gives you an added channel count to the inputs so that you can actually have a total (stage and local FOH) inputs of sixty four. The outputs are still capped at 32, which is certainly ample for running a full compliment of mixed wedges and IEMs.

Pro’s

I love the snapshots page from this console. Of the three I have reviewed (the Midas PRO2, the Digico SD9, and the SC48) the SC48 has the most strait-forward automation and snapshot interface. The ability to click and drag elements within your set-list to different points in the order of the set is amazing. If the band decides to change up the order of the songs, you just click and drag them into a different order.

I also really liked the Fine button on the console, which makes all your adjustments fine-tuned. The fader motor drags on you slightly giving you the ability to make subtle changes within a few tenths of a dB, and the encoders make changes at about a third the rate of change as normal allowing you to really nail an EQ frequency to notch out just one specific frequency or fine tune the attack and release times on your dynamics. The gain guess feature which I didn’t get to use is great because double tapping on the mic pre knob tells the onboard computer to guess the proper gain amount of a channel by sampling about 5-10 seconds of its incoming signal and setting the mic preamp accordingly.

Con’s

If you’re like me and want to jump into the deep end with plug-ins I think their use makes you myopic in a dangerous way towards mixing. You lose sight of the big picture when mixing that way. I find it more of a distraction than a tool to use. I think plug-ins are best when they are incidental use for special effects. The misguided thought that you can have an LA-2A, Fairchild compressor or API plug-in on every channel is also dangerous. And the plug-in are expensive as well. You can max out the DSP really quick with a few big plugins, and if your plug-in rack crashes (and yes this does happen) you are totally hosed. That’s because while the engine will still pass audio, you are locked out from making any changes, and rebooting the plug-in rack you lose audio temporarily.

I find the rotary encoders a bit clunky in terms of feel and how they move. I also found my workflow (when I wasn’t busy tweaking plug-ins) a bit slow.

The Digico SD9

noreply@blogger.com (Anonymous) — Sat, 17 Mar 2012 02:18:00 +0000

These past few weeks have been quite action packed me. I drove to Long Beach to demo the Digico SD9. I should state that i didn't actually demo the SD9 as they didn't have one available. however we looked at both the SD8 and SD11. All of these consoles run the same software. The real difference between these consoles is the work surface, and how many features are available with the DSP available.

The I/O
With the SD9 you can connect up to 2 D-Racks. This will give you up to 64 remote inputs and if you also fill the unused output card slots with analogue or AES/EBU cards you will have 32 remote inputs. the console also has a few AES/EBU input outputs, and eight analogue inputs and eight analogue outputs. This could give you a total of 72 inputs and 40 outputs.

The Software:
I'll be honest. When I first downloaded the software I was very confused by it. It is not laid out in a way I find intuitive. For me, that's going in and setting up the patchbay first, where the inputs and outputs are and how they would enter and leave my console from other equipment in our system. It also means setting up a basic template of what a typical Sunday morning would look like mapped out on this console. I would create all various mixes, groups and matrixes. And it took me a long time to use.

It has a very clunky Windows 95 feel about it. And what's more is that they made it so very cramped. The image to the left shows input channels 1 thru 12. Packed into each channel display shown to the left is several things. First is the channel input. From the channel input you also select where channel 1 sources it's audio from, either a physical connection on the I/O or an internal group or matrix. Then you have your high and low pass filters, your first insert point, your EQ section that also has dynamic EQ's that can be applied to bands within the EQ. Followed by that is the compressor, then the gate, then you have the second insert point, then you have visual support for your first six auxiliaries. Then you have your panning section, finally you have your digital scribble strip.

Virtual Soundcheck:
Running Virtual Soundcheck on the SD9 is so simple. I would grab the Digico UB MADI which converts MADI coming from the console to a USB that you plug into your computer and your entire desk becomes a giant recording interface for the first 48 channels of audio. And because it's a simple USB interface you can use your laptop, PC, Mac, whatever. And then I would probably purchase the $80 license for Reaper recording software. Playing the audio back into the console is simple by pressing the "Listen to Copied Audio" which feeds the audio back into the channels. The UB MADI will set you back somewhere over $1000. But I think that it's a huge bargain for recording 48 channels!

Plugins:
Like other consoles, you can integrate Waves Soundgrid. It's expensive but having the option is a perk.

The Pros:
I do think this is the future of digital mixing. The touch-screen makes interacting with the console a two handed approach that cuts down on some time. I really like the ease of making virtual soundcheck work on this console. The use of MADI integration is very wise. You can also change what you need this console to do very easily and free up DSP for added effects engines. You have touch-capacitive sensitive faders, eight segment metering on each channel. The channels themselves can be flexible to be either mono or stereo. Hence the name "flexi channel." All of your bus outputs are flexi as well from mono, stereo or LCR. You also have two separate solo busses and eight macro keys.

The Cons:
My opinion of the software is that it is very un-intuitive. The GUI has a sort of chunky Windows 95 feel to it. I am not fond of the console layout. You are still working in layers, And given that the layers are split across the two twelve channel fader banks, you essentially have eight layers to navigate. You also don't have it set up in a way where the 12 channels you are working on aligns with the touch screen. They are always offset. I get the feeling that my volunteers would be confused by working on this console and in the software.

In all you may think of this review that I didn't like the console, but that's not true. i did like it, but I don't think its friendly for a "volunteer run, staff supported" ministry, which is the production philosophy of our church. I need to be able to trust that my volunteers can walk up to this console, set up the console for the needs of the particular week and lead the band through sound check and run a show. If I were working in a church where I and perhaps 1 or two volunteers were mixing Sunday morning services and all special events in our main room I would use this, but that's not the situation I'm in and I have to subordinate my personal preference to our vision and needs.

At First Glance: The Midas PRO2

noreply@blogger.com (Anonymous) — Fri, 02 Mar 2012 19:16:00 +0000

Right now as a church we are starting the process of moving to a digital desk. That means I get to drool on lots really expensive gear. Or, so I thought.

Turns out that our budget is a bit less than what I would have liked for a console. Our price range is in the sub- $30,000 price tag. Mind you, there are still several great consoles to be had in that price range. So based upon several conversations I have had with our leadership about what we want to do in production over the next 5-7 years I started generating a short list of candidates to demo. Then, talking to some of my other audio geekery friends they said, “if you’re checking out this product you should also look at that one too.” Which has added more to the list of candidates. I took initial looks at some products and then eliminated those that don’t seem to be what we’re looking for. Today however, I’ll be talking about the Midas PRO2 from the demo I did recently.

The Scalable I/O

Possibly the biggest selling point on why I love this as an option is that it doesn’t use a fixed I/O. The standard PRO2 comes with a DL251 I/O rack with 48 inputs and 16 outputs. Local I/O on the console has another 8 inputs and 16 outputs. But if you need more inputs and outputs you simply add more. Plug in a CAT5 cable and buy a DL451 with another 24 inputs and 24 outputs. The I/O racks connect to the console via AES50. So what you can do is put I/O where you need it, and up to 100 meters away. I could put my 48 in 16 out on the stage and let it be my input for microphones and output for IEMs. I could also put a rack in our electrical room to have the inputs interface with our tie lines and the outputs can be the shortest possible cable run to our amp racks. Say goodbye to all those copper snakes. And the entire desk runs at 96kHz, with a 40-bit floating resolution.

And you can keep expanding your I/O all the way up to 156 inputs and 166 outputs. WOW! By that time, you might need a bigger console just to manage all those channels. But because Midas shares all the same I/O racks, you could just trade in your PRO2 (leaving all the I/O architecture) and drop in a PRO9 or XL8. This also means that it will still sound like a Midas. The console also features dual hot-swappale redundant power supplies. Not that you really want to try that feature during a Sunday morning. But knowing that there is that safety factor built in is very comforting.

Console Layout

On the Midas you have 6 POP groups (or Population groups) that by tapping the POP group it temporarily places the channels within that group just to the left of the VCAs. VENUE users know this as VCA/Group Spill. But unlike the Midas, the VENUE consoles have 8 groups.

One of the things I don’t really like looking at consoles in this price range is that you don’t have 24 channels on the surface; you have 16. On the PRO2 however, there’s a little button that expands the inputs channels onto the VCA faders so that you can mix 24 channels at a time. I also found out the other day that each Aux can also be a group bus, or a mix minus bus. In today’s world everyone wants to do is control the console by a laptop or iPad through a VNC client. Midas has submitted it’s own iPad app that is designed for your finger and not a mouse.

The Software

I’ve spent some of my spare time the past few weeks working with the console offline software. The Midas software is very a very intuitive user interface that seems to work well. What’s more is that the software is the same for all their digital desks. The man I did the demo with stated that he could take show file from his PRO9 or XL8 and put them on the PRO2, provided that the channel count isn’t over on the PRO2. For some reason I find the software looks less cluttered than the Avid software. Not to mention that you actually have a SAVE button in the file window that makes it simple to know how to update your scene or snapshot. In the D-Show software you right-click on the showfile you’re working in and select Overwrite. Seriously?? No save button that just lets you save/update the scene you’re working in? On the Yamaha desks you have the Save button right on the desk without having to dig in the software. I’m still not sure how to save my work on the Digico.

Virtual Soundcheck:

The SC48 will allow you to record up to 32 channels into ProTools with the new D-Show 3 software. For this you need a computer, FireWire cable and install the supplied ProTools LE software and some external hard drive space for all those audio tracks. With the Midas you would buy the Klark Teknik DN9696. It’s a 96-channel 96kHz audio recorder that integrates directly with the console via AES50. Now where as the Avid will cost you a little over $1000 for a computer, mouse, keyboard, screen and external hard drive space the Klark Teknik DN9696 will set you back a whopping $14,000. I was astonished to be honest when I saw that number on the price quote. It’s almost as expensive as the PRO2C. One possible workaround for this would be to buy the Klark Teknik DN9650 Network Bridge to convert the AES50 digital audio to MADI then run it into a JoeCo MADI recorder. You get 64 channels of recording and playback instead of 96, but the cost will be a little less than half the cost of the DN9696. With the SD9 you can grab the UB MADI and any computer with a USB 2 connection and with Reaper recording software you could record 48 channels of audio for less than $2000.

UPDATE 3/5/12: After looking around I actually can do virtual soundcheck cheaper than listed above. I would still purchase the DN9650 network bridge, but would buy the RME MADI PCIe Card. I would then buy Reaper recording software to record the MADI channels and stream the outputs back to the desk. This option cuts the cost of virtual soundcheck to about $3000 over the cost of the console. END UPDATE

Plugins

The console will integrate with Waves Soundgrid so that you can use any Waves plug-ins. But I’m somewhat ambivalent towards plug-ins anyway. If you can’t get your mix sounding good with just the channel EQ, expander/gate, and compressor, you should spend more time focusing on your mix rather than messing with plug-ins. Seeing Lexicon 480L’s, TC electronics, and Yamaha REV5’s and SPX’s on the shelf where I did the demo I asked the man who was demonstrating the console what effects units he takes on the road to mix. His answer, “none.” He said he uses all the onboard Effects units. The onboard compressor has four different modes: corrective, which is a peak sensing compressor with exponential attack. Then there is Creative, which is peak sensing with a linear attack. Adaptive mode is like most RMS sensing compressors, then there’s vintage, which adds some coloration to the compression.

The Price

First off, there are 2 options for this console, the PRO2 which has 28 faders on the control surface. Then there’s the PRO2C, which is the same console but minus 8 of the input faders. So it has 20 faders on the console. I was shocked how much power and versatility was packed into this console. The PRO2C with the DL251 rack (total of 56 inputs and 32 outputs) can be had for a little under $17,000. The PRO2 with the DL251 rack (again, 56 inputs and 32 outputs) can be had for under $23,000

Pros of the Midas

The Midas can completely scale its I/O to meet your needs. Auxiliaries can be configured to be a mix, group buss, or mix minus buss. All of the inputs and outputs can be color-coded to make it easy to group channels or outputs. You have two solo busses. Variable high pass and low pass filters on each channel. Matrix outputs or buss outputs can have either a 6-band parametric or 31-band graphic EQ. You have six free-assignable rotary encoders I forgot mention. Six unique mute groups. Monitor send flip to faders. The rotary encoders are touch capacitive-sensitive. You can mix in any format from mono to 5.1. It has its own app for VNC control. It shares the same I/O as the XL8 so it sounds like a Midas! You can change processing order on channels… There’s way too much stuff to list.

Cons of the Midas

The show filing and presets are perhaps too comprehensive for a novice. The console wants you to be intentional about what you are doing and saving your work. Every parameter can be scoped and saved into the show file. This can be daunting the first time you do it, but once you have you patching done and global settings done, you won’t need to touch that page.

The price of doing virtual sound check on this console is an optional extra. But then again it’s the same with the SC48. You still need a mouse or trackball and a computer screen that is not included with the SC48 console. You also need the AO16 card (roughly $2500) to bring the I/O to a maximum of 32 outputs, a computer running ProTools, a screen, mouse, keyboard and some external storage space. But even still, the price on the Midas is ridiculous if you do it the way Midas and Klark Tecknik want you to do it. $14,000?? I know it’s a 96-channel recorder that records broadcast WAV files at 96kHz but I’ll take the other, cheaper 64-channel solution I outlined earlier.

I have the SD9 demo and Avid demo packed into next week’s schedule. It will be a hectic week but I can’t wait to get to know these consoles and find the one that meets our needs. I’ll be posting later on my opinion on the other consoles.

(The Real) Three Way Mic Shootout

noreply@blogger.com (Anonymous) — Wed, 01 Feb 2012 18:11:00 +0000

So, as promised I have the three way shootout of the three way shootout between the Sennheiser HSP 4, Countryman E6 and I threw a wireless SM58 in there because it's a microphone that all of us audio guys are familiar with in it's frequency response. I apologize for the less than perfect video quality Part of the reason is that I'm not that great at finding the right settings for video file compression to get it look good while still keeping a smaller file size. Though it sounds like there's some slight high frequency detail lost, this is the typical setup for our video encoding to web for our Sunday morning services. I highly recommend using headphones for this as the differences are not noticeable on computer and laptop speakers.

My one caveat to this that slightly corrupts this test is that I forgot to bypass the compressor that hits our video inputs so, yes the sound of the mics is not entirely natural, but our typical compression is only about 2dB to 5dB of gain reduction when our pastor starts to get excited and really project so it's not that heavy on this video. My own personal opinion is that the E6 over accentuates the nasally tones in my voice's upper-midrange detail. Sort of 1kHz to 3kHz area. And there's not much natural fundamental frequencies in below 300Hz on this mic that sound good. My voice is slightly nasally to begin with but this mic leans me more to sounding like Gilbert Gottfried. You know, the guys who voices the Aflac duck? No offense Gil.

The Sennheiser HSP 4 sounds very natual. There are fundamental frequencies that add body. I was expecting a lot of low frequencies from this mic being that it has a cardioid pickup pattern and would accentuate proximity effect, but it didn't. The nasally quality is not there as well. The sound is more neutral, and I like how the high frequencies sound on this mic. In the room on Sunday morning I loved how clean the mic sounded. Rather than Pastor Josh sounding slightly synthetic and that he is wearing a mic, he sounded like he normally does when we just have a conversation which was a huge win.

Is this a Microdot connector??

The SM58, with it's big dynamic capsule sounds really in your face and intimate, lots of low mids. But given that I planted the microphone on my chin and spoke directly into the capsule, it would sound that way.

Working with the mic on Sunday morning I loved the sound of the mic first service with the EQ bypassed, but one of my volunteers Chris, a former recording engineer felt the desire to tweak the EQ to make it sound better. Looking over the settings on the console between the E6 settings and the Sennheiser settings we did about half the work of getting it to sound right as we would on the E6.

The EQ settings for the E6 channel has a 100Hz high pass filter and then the EQ is set as HF Shelf: -9dB at 20kHz, HMF: -9dB at 700Hz, Q= 1.4, LMF: -12dB at 220Hz, LF Shelf: -2dB at 160Hz. Crazy EQ setting yeah!? My thought too. There's gotta be something I'm doing wrong. The E6 has several capsule filters that change the frequency response of the capsule. I have the neutral filter on the capsule and even still the HF shelf at 20kHz is there because even with the natural capsule that doesn't boost high frequencies we have issues of feedback above 12kHz. And those are feedback frequencies that are very painful. The two mid band EQ's are what I'm shocked by. Why do I need that much EQ to make it sound good or neutralize feedback frequencies?

UPDATE: what I forgot to mention is that getting this mic to test for a few days meant that I got to torture test it with a few voluneers. So I put it on and cranked it as loud as I possibly could. I figure the house level was around 75-80dB. And then only when I went into the house standing in front of the PA and talking (and annoying the band members who were arriving to set up for rehearsal) did I hear the slightest beginnings of feedback. But that was standing in front of the PA!! END UPDATE.

I'm trying to arrange for a demo of the DPA 4067. For you DPA users I was told that the connector pictured above is the MicroDot connector, I have never seen one, is it? Still no word on when the DPA demo will happen but I'll keep the posts coming on what is going on. Stay tuned!

Three Way Mic Shootout

noreply@blogger.com (Anonymous) — Thu, 26 Jan 2012 16:24:00 +0000

I set up a three way demo of the microphones between the Sennheiser HSP 4, the Countryman E6, and a wireless SM58 beta. One of the mics gets jealous. Watch the video to find out what happens.

Headroom, Feedback, & the Keebler Elves Part 2

noreply@blogger.com (Anonymous) — Tue, 24 Jan 2012 04:35:00 +0000

The quest for better sound for our pastor has me landed on two product demos. The first one is the Sennheiser HSP2. However the HSP2 wasn't available for demo, but I got the HSP4, which is the cardioid version. The reason I chose this is that it uses the MKE Platinum capsule, and we have an MKE clip on lavalier microphone that we used to use and it sound very good. Then we moved to the E6.

There are several benefits of a headworn mic over a shirt lavalier. As the pastor looks down to read his notes or a Bible passage, the lavalier will pick up a more direct detailed sound because he is speaking right into the mic. Then he looks up and it sounds different. If he moves his head too much to the left or the right wearing a lavalier, the sound is also likely to change. And not just in tonality but in volume as well. So the inconsistency is not good. I don't have a great video setup, it's called an iPhone and iMovie. But I wanted to show you guys some things about this mic that you wouldn't naturally see if you saw this online.

The one thing I mention in the video that I'm still not sure is accurate (please correct me if I'm wrong) but the Sennheiser Rep told me the detachable cable uses the Microdot Connector. However looking online I think the connectors look slightly different. This one presses into place, and the DPA Microdot looks like it may be a threaded connector. Are they the same?

Sennheiser HSP 4 Preview from Jeremy Blasongame on Vimeo.

For me wearing the microphone, I find that I tend to shoot "D" and "K" sound out the side of my mouth. But a combination of moving the microphone back a centimeter or two and the low frequency roll off helps out. In the video I mention that the sound quality is much better on the Sennheiser. In fact, what I did was neutralize the entire EQ section and the only thing that was active in my tests was a 80Hz roll off to keep low frequency plosives out of the subwoofers. After walking the room I added 3dB at 1kHz. The difference between that and the E6 is huge. My high-mid band on the Pastor EQ is set to -9dB at 700Hz and the low-mid band is set to -12dB at 200Hz. A huge difference. And I was able to crank it before I started to hear feedback.

In the past I didn't think to consider a cardioid microphone but I'm actually giving this one serious thought because it sounds so natural. I have the mic until next Monday so I'm going to have my pastor wear it for this Sunday and then post short video snippits of A/B differences between last week's sermon of him wearing the E6 and a sermon of him wearing the HSP4. Obviously I'd go with the beige version of the mic which adds the -3 designation to the product code (so ordering it would be the HSP 2-3 or the HSP 4-3). I think the biggest difference from the two microphones is that the HSP 4's capsule is more visually obvious. It is 8.4mm wide, which is actually quite big. the HSP 2's capsule is almost half the size. But without having the HSP 2 available for a direct comparison, that is the only thing I can comment on.

Again, thanks to Sennheiser for helping arrange this demo. I hope this is a great test, and I'll get back to you guys with info on how the test went.

Headroom, Feedback, & the Keebler Elves

noreply@blogger.com (Anonymous) — Fri, 13 Jan 2012 17:55:00 +0000

The last time I blogged it was 2011. Christmas is a big reason for it. Getting engaged was another.

One of the solutions I have been looking for is something that would help suppress feedback on our pastor's microphone. He loves to engage with the audience and so he stands on the edge of the stage. We're always wondering when he's actually going to fall off. ~~And catch it on camera. To post to YouTube. And then re-tweet~~.

Exactly fifteen feet above his head is the PA and when he turns his head in just the right angle you will hear the beginning whispers and groans of feedback. He's wearing the Countryman E6. And I'm not a big fan of the sound quality from it. But it's not at one frequency I could attenuate slightly. It varies with his position and where he's at. There's some feedback above 12kHz, some at 800Hz, some at 270Hz and some at 150Hz. Bottom line it's too much to start hacking away with notch filters and make him sound unnatural. To counter this problem I've simply turned his mic down a few dB. The only problem with that is his volume isn't where I'd like it to be. If you're sitting in the back of the room or with me in the tech booth it's parked around normal conversation volume, about 60-65dB-A.

I'd like to get that number 5dB higher at the back of the room so that's had me looking at different ideas which brings me to the subject matter today the Rupert Neve 5045 Primary Source Enhancer. I watched a demo provided by the Yamaha rep at a local dealer. He brought a microphone, a speaker and inserted the 5045 to a channel on a console and he started with all the knobs at twelve o-clock and hit the insert button. Instantly the feedback was gone. This made one of my eyebrows raise up. The Yamaha rep raised the volume a few dB higher and fiddled with the knobs and the feedback started to go away again. I was stunned, and my other eyebrow followed the first. And thankfully I was able to take it with me to try for the weekend. I did all sorts of things to test it out.

Just for kicks I tried to figure out what is going on under the hood. The rep I talked to at the demo compared it to a "soft gate." A soft expander would be more accurate I think. Later when I got to test it and try to figure out how it worked I tried it on some CD music and yes it is sort of a soft expander. There are three knobs on the device, including a blue illuminated Engage/Bypass switch. First is the big red knob that selects the different time constants for the attack and release. There is also a RMS/PEAK switch that changes the time constants. For our pastor I liked the sound of the RMS. The middle knob is a variable threshold. A LED next to the knob helps gauge the threshold so that you are not processing too much and chopping away at the pastor. The third knob is sort of the depth knob it's range is from zero to twenty and the rep boasted that if you set it up just right you can get almost 16-20dB headroom. I only squeezed about 7dB more headroom from the mic, which is great. One also has to consider the speaker system, the mic you're using, and other factors to how well it will perform.

My boss was in the auditorium while I was trying to make microphones feed back and when I pressed the button to engage the process and all the feedback went away it was magical. She even said "wow."
Sunday morning was great. I loved hearing our pastor a bit more from the system and there is some magic to the Rupert Neve name, the device with the processing bypassed just sounds good.

My Verdict: It has the magic as the Keebler Elves yes, but there are a few things I don't like about it. First is the price. It's around $1600. Yes there is two channels so it's only $800 per channel but that still on the edge for me in price. For that price I could buy a much better head-worn microphone for our pastor. I'm looking at trying the DPA 4066, and the Sennheiser HSP 2, not as well known as the DPA but features the MKE platinum capsule, which sounds awesome.

I know it may sound like I'm nit picking on price but I really do love this piece of equipment. Special thanks to Yamaha for loaning the equipement and to Apex Audio for helping arrange the demo!!

Great Drum Tones: The Cymbals

noreply@blogger.com (Anonymous) — Fri, 02 Dec 2011 18:46:00 +0000

Continuing my series on getting drum tones I want to talk about cymbals today. Many live sound engineers will not understand this problem, with the exception of working in small clubs or in churches where the stage volume of musicians can be an issue.

Of the progenitors of stage noisiness, the drums are king. In many churches its such an issue that drum isolation booths from companies like Clearsonic or Perdue Acoustics have become the staple item. I know of some churches that build a complete “drum condo.” It’s a fully enclosed room on the stage, with a door and you can stand outside the room and carry on a normal conversation without having to raise voices while the drummer is pretending he’s John Bonham. It wouldn’t surprise me to find a Christmas wreath hung on the door of the drum condo this time of year.

Thing is, though, drum booths and drum condos and even just drum shields are little more than a necessary evil, they are visually intrusive, and the added reflections impact the sound of the drums. With some drummers using a shield I notice that I have cymbal bleed coming into tom mics, and even the kick. So then I spend more time as a butcher than a mixer, hacking and slicing away with the EQ trying to minimize the cymbal bleed while giving clarity to the drum microphones.

A while back I found a YouTube video of the Zildjian Gen16 cymbals and showed itto one of my drummers. He was very skeptical but willing to try it. We’ve been searching for a better solution for drum stage volume issues and I came across a video of the Zildjian Gen16 cymbals. On paper, and YouTube, they looked good. But the compressed low quality sound of the video meant that if we wanted to see how good they really are, we needed to try them out. So we bought them with a 45-day money back guarantee knowing that if we hated them we only had to pay for shipping them back.

Let me be very clear: THESE ARE NOT DIGITAL CYMBALS. I can shout into the microphone capsules under the cymbals and have them amplify my voice into the PA. They are still cymbals, but about eighty percent quieter.

The way they work is that they have a capsule under the cymbal that has two microphones inside that amplify the cymbal. There is a small breakout cable that runs to the cymbal capsule. The cable connects to to the DCP (Digital Cymbal Processor) that has preset EQ curves much like an acoustic guitar that has a preamp and tone shaping capability built into it. So you dial in a preset EQ that you like and get playing, and the presets are made to sound like other cymbals. There are twenty presets for each cymbal available. This means that you can make the cymbals sound really bright and sparkly, or really dark if you want and everything in between. So one preset will be a sound much like a K series cymbal, and another like an A- series, or a Z-custom.

The 480ae kit we bought came with the DCP, the cabling, 14” hi hats, an 18” crash cymbal and a 20” ride. Zildjian also has a complete lineup of different size crashes, rides, splash cymbals and china’s for drummers to complete their drum kit how they want it. And for some of you crazy drummers who like to use small crash cymbals on your hi-hat, go for it. It will work. The key to these is spending a bit of time finding the right tones out of the cymbals you like. In the same way that you go to a music store and take a pair of drumsticks trying different cymbals out to find one that matches the tone you are going for you need to sit with a good pair of headphones or in-ears and just find out which tone presets work for you.

When they arrived in the mail I called one of my drummers and he came over to play for an hour or so to get tones out of the cymbals and try the drums without the shield. Probably the coolest thing in trying these cymbals has been that because it’s so much quieter, I don’t need the drum shield. The musicians are asking for much less kick and snare in their monitors and I was able to neutralize some of the radical EQ work I did on the toms to kill the cymbal bleed with the drum shield. The result is that the kick, snare and toms sound much more natural, and both of us were very happy with the results, him as a tone-picky drummer, and me as a CDO (that’s OCD in alphabetical order) FOH engineer. The image above shows how much cleaner our stage looks for Christmas without the drum shield.

My Verdict: It’s not a product for everyone. For the absolute purists who love the sound of real cymbals this will be something of an abomination. You do need to spend a bit of time finding the tones you like. The first week we tried it we weren’t as satisfied as the next week. But for many, including me, this is an amazing solution. The kit we bought runs about $1000 street price, which is very reasonable, and add-on cymbals are less than the price of a K-series crash cymbal. Yes you have to deal with a bit of bleed from the snare and toms, but it’s manageable. And remember they’re not digital cymbals playing back a recorded sample, they are real cymbals. I have been able to get away with taking down the drum shield, which leaves our stage looking clean, and makes the communication with the drummer much easier. As a result the stage volume is lower and I don’t think we’ve had any volume complaints centered around the stage volume of the drums in a few weeks, which is a huge win! In fact, we’re planning on purchasing another Gen16 crash soon.

If you're local to the Southern California area and are looking to purchase these send me an email through my profile or DM me on Twitter. We can try set up a time you can come over and play them. Oh, and you will have to bring your own drummer, because I can't play.

Great Drum Tones Part 1

noreply@blogger.com (Anonymous) — Sun, 20 Nov 2011 23:23:00 +0000

I'm starting a series of posts on drum tones. Being a bass player turned sound guy, I'm quite picky about drums and bass in my mix. So in the following series of posts I'll talk about microphone placement and choice on the drums followed by mixing techniques.

The first thing to remember is that the drums as a whole are an instrument. Yes the snare is one of those instruments but I have never heard a song that only used a snare as a percussive instrument. Usually it's several elements making up a drum kit. With that in mind it can be challenging to get a good close mic sound from individual instruments within the drum kit. I've found that 80% of the sound that comes from the drums is the drum itself. Wood drums sound different than metal drums as well. Wood drums are typically Maple multi-ply wood, while metal cymbals are either brass or steel. For my preferences I tend to prefer wood snares over metal snares. Metal snares, especially brass, have a distinct resonance or "ping" sound to it that I find is often dissonant to the key of the song and I waste of band of EQ to try and neutralize that resonance. With wood snares I find the overall tone may be a bit darker but there is a cleaner pop sound. There is a lot about the snare sound that is out of the hands of the audio guy. Some drummers like single ply heads, while others prefer multi-ply heads. Some don't realize that this choice will effect tone. But lets assume that the drum has a head that is in new or good condition and everything is tuned properly. Not that I'm slamming drummers but most of the drummers I have met don't know how to tune their drums properly and it just creates more work for me hacking and slicing at the drum tones when I go to EQ. Then I've worked with other drummers who bring in great well-tuned drums and have very little work to do because they already sound great.

The other twenty percent of drum sounds obviously comes from what you and I do with the microphone and all those knobs on the console. Starting with the microphone it's very common to use an SM57. I use the SM57 beta because I got tired of drummers popping the plastic head off the standard 57. I'll place the microphone about a fingers width above the rim like the image pictured here aiming where the stick makes contact with the head of the drum. For this drummer it happens that it was the center third of the drum. For most guys this is great. I like to use a second mic underneath the snare to get the sound of the beads of the snare and layer that in with the top snare mic.

Typically for a snare bottom microphone I will place it about two inches away from the bottom of the snare facing up towards the snare beads. If you're using a bottom snare mic you will need to check the phase between the top and bottom mics. Sometimes if you're using a 57 on top and a condensor on the bottom you'll be OK, but more often than not there is some phase cancellation between the two mics. I personally like the SM87 as the snare bottom mic. It's a cheap alternative to the AKG 451 that I loved to use as a snare bottom mic in the studio. In live sound close mic'ing the drums gives a lot of clarity and up-front emphasis to an instrument especially with bleed from other instruments like guitar amps or bass rigs.

Next time we'll talk about the kick drum and hi-hat. Happy mixing!

A Better Way To Recruit Volunteers

noreply@blogger.com (Anonymous) — Tue, 11 Oct 2011 19:10:00 +0000

The problem with being in tech at a church is that you are IN tech. The nature of our work means that we aren't afforded the time to go socialize with prospective volunteers to recruit them. At least, I don’t have that kind of time. And I’m not the sort of person who can easily walk up to someone who looks new and say, “what would it take for you start volunteering in the Sunday morning production?" Bulletin announcements never work, announcements from the pastor generates willing bodies, but few actually have the skills or what it takes to be effective. Which leaves you scraping, begging and endlessly bribing people to help you in tech. After a few months of that I said to myself, there's got to be a better way!

So, a few months after I came on staff I sat down with my supervisor and the pastor and talked about the need for systematizing volunteer recruitment. After all, tech isn’t the only area that needs volunteers. It seems there is a continual need for people assisting guests in the parking lot, children’s ministry, ushers, greeters… you get the idea. The bigger problem is that when we only tap people our relational networks on the shoulder to help us, we over extend people who are currently serving in sometimes two or three areas. And we miss out on empowering new people who walk in the doors of the church looking to get involved and contribute.

So my supervisor and the lead pastor created a half sheet questionnaire that is used in our quarterly “Discover Sunridge” class. It’s a class where we explain why we exist as a church and unpack our vision and values. We explain that we don’t have formal membership, but that if people want to really feel like Sunridge is their home church they should find relationships and a role. Relationships come through our small group structure where people do life together and share time in the bible and in prayer caring for each other. Roles come in the form of volunteering. We want people to contribute in some way, not just lurk on the fringes. The half-sheet questionnaire asks whether they’re a people person or not, when they are available to serve, and what skills and passions they have.

For example, you need to be super outgoing and a great people-person to be a greeter or to assist guests find their way around the building and use our child check-in system. Tech people need to have lots of patience, a cool head under pressure and a team-player mentality. Those are two completely different kinds of people we’re looking for, but both are very valuable and necessary for the church.

The sheets are reviewed for the best ministry fit based upon gifting and time available, then they are placed in the appropriate staff member’s box and we are told to make contact with that person within a week to talk about volunteering. We arrange to sit down with the perspective volunteer (a great excuse to grab Starbucks) and vision-cast why we do what we do and ask them questions that really assess whether this person is right for this ministry. If they seem to have the right skills or the right attitude and willingness to learn, the training begins. If not, we try and help them find a ministry (and ministry leader) that is a better fit for them.

Alexis, Bryce, Jacob, Jamon Jay, Joe, Jolene, Mike, Paul, and Robert. I guess I attract all the ‘J’ names. Over the past ten months these are the ten people who have come from that recruitment system into not just audio, but cameras, video direction, lighting, and ProPresenter. In all fairness, one has decided that tech is not for them and is serving elsewhere, but that still means that nine out of ten volunteers is 90% effective at finding the right people to work in tech. Right now I can’t take on any more audio guys. Not even if they used to work in a recording studio and have mad mixing skills. I think I may have too many audio volunteers as it is. Some of my guys aren’t getting enough time on the board for consistency. I still need one more person in lighting and one or two more cameramen but the next Discover class is in a few weeks, and I’m praying God will provide for those needs, especially as we are growing in our Sunday morning production. Right now, if I'm just technically directing and not mixing, it takes ten people to make a Sunday morning service happen.

The Discover your role card is almost overly simplistic, but we want to keep it short and simple. Besides, ministry leaders are the best people to assess whether a person can be effective at serving in their area. Tech Directors who struggle to find the time to recruit volunteers should forward this post to their pastor or supervisor. And if you’re a pastor who got this post emailed to your inbox, it’s a win-win situation that has helped us with volunteer recruitment in every area of the church. And I'd be willing to talk more in depth about how you can use our system for the benefit of your church, you can find my email on my profile page.