My Parallel Processed Bass Rig Part 1: Why Parallel Processing?

I’ve wanted to write this post for a long time. It’s a sort of reply, or alternate take on Johnny Ragin, AKA, Worship Sound Guy’s YouTube Video, “My Super Weird Trick for HUGE BASS.” I’d like to take the approach from a bass player’s point of view. If you’d like, you can watch it below. If you don’t want to watch it, keep scrolling.

After looking at the date, I realised Johnny posted this video a year ago. Where did the time go? It’s a cool video that explains a lot and does a great job of explaining it in terms a layperson could understand. A layperson is a sort of kinder gentler way to describe a novice. Johnny shows how to do this in Pro Tools, THEN he turns around and does it on a Behringer X32. The cool thing is that this trick is pretty easy to pull off on any digital console, and can be super simple in analog world too.

So What is Parallel Processing?

I would define parallel processing as splitting an audio signal into two or more paths with separate processing applied to each signal path. If that sounds scary to you, chances are you’ve done this before and not even thought about it. On an audio console any time we use an auxiliary send to add reverb to a vocalist, we are adding a parallel processing path. The vocalist’s voice goes straight into the console’s channel strip and splits at an aux send. The voice continues through the channel strip; equalisation applied, and the level controlled via the fader. The aux send sends the signal to a reverb processor. That reverberated signal gets mixed back into the console on either another channel, an aux return, stereo return, or something like that. Usually, if you mute the reverb, you can still hear the vocalist.

But Why Though?

Why not parallel processing? Everybody else is doing it. (Peer pressure is not a good reason to do something, by the way.) Chances are, every single bass you’ve heard on a recording is parallel processed. I don’t have stats for that, but if I had to guess, I would say at least 95%. I would also venture to guess that if you’ve been to a concert, especially nationally touring acts, they also process their basses in parallel. Here are a few reasons why.

Why It Works for Clean Bass

Clean bass tone is the anchor of a lot of music. . Several bassists such as Marcus Miller, Tony Levin, Nathan East and others have built entire careers off of clean tones and virtuosic playing. Some people would define this as a "hi-fi" tone. You can get a lot of low end out of a clean rig. So you may be asking yourself, "what's the problem? Don't we want low end?” The answer is a resounding yes.

The problem with a clean bass tone is, that, believe it or not, it can get lost in the mix. The fundamental note of the bass drum can compete with it. In modern music basslines played by synthesisers, or coming from backing tracks fight with the bass guitar for space in the mix. 

We can work around this. Add a second bass channel with a distorted amp, or pedal. Mix back and forth to taste. Distortion adds harmonics. The harmonics start to move into the midrange frequencies, which can make the bass easier to hear and differentiate from the synth or backing tracks.

Why It Works for Dirty Bass

Distorted, or fuzzy bass can sound really cool. You’ve heard it. Pretty much every rock song of the 1970s has a gritty sounding bass. Muse is an excellent example of a band that uses distorted bass well. They’re a three-piece band, so sometimes the bass player adds distortion to the bass to sort of taking over what a rhythm guitar player would do. By taking over the rhythm guitar parts, the bass player frees the guitar player/vocalist up to play leads and sing.

Unfortunately just adding distortion or fuzz to bass can sometimes eat the low-frequency stuff that bass usually plays. It can thin out the tone and make it almost sound like another electric guitar. That’s great if that is your goal. If not then splitting the bass signal and running it through a clean amp, and a distorted amp can bring that missing “oomph” back. If you need more “oomph” simply blend in more of the clean amp.

Why It Works for Other Effects

The reason parallel processing works for basses with effects on it is the same reason it works so well for distorted or fuzzed bass. Any time we add an effect like chorus, flanger, phasers, envelope filters, or whatever, we can lose some of that oomph that our instrument has. This is especially apparent in the low end. By splitting our signal and always having a straight, “clean” tone available, we can negate that loss by mixing what we need back in.

My Parallel Processed Bass Rig Part 2: The Gear” is up next where I give a little information about the gear in my bass rig.

Ghost in the Machine: Me Vs. a Dante Network

A few months ago, I think it was early March, I went to help a church get some problems with their Dante audio network sorted out.  (Yes, I need to write more often- don't worry,  Mom says the same thing.) They currently have a Yamaha M7CL with two Dante-MY16-AUD cards connected to a Cisco switch, which is connected to another switch in their production room. The two Dante cards handle 16 channels of audio each, so that they can get all 32 channels to their recording room. Their production machine is a Mac Pro running Apple's Logic software.

In full disclosure, let me make the following statements:

  1. I should have payed more attention in NET 125 Networking Basics. But the material was super dry and let's face it.  That's not the sexy side of playing with all of this awesome audio gear. Am I right?
  2. I know people that have fairly complex Dante networks that are working beautifully. For example there is a local college that is cramming audio through their regular infrastructure. Meaning there are multi-channel strings of audio swimming in the same stream with college students streaming Downton Abbey, The Walking Dead, Mad Men, Breaking Bad and Walker Texas Ranger. It works.
  3. Dante works beautifully, and is almost completely "plug and play" on consoles that run Dante natively (that don't need an expansion card) like Yamaha's new CL series consoles and Rio stageboxes.
  4. I typically dislike thing that take a lot of effort to set up. Let me get to what I came to do as quickly as possible. Which is usually mixing.

The Problem

In this particular case, the problem was that the same audio data would show up on channel 1 & 17, 2 & 18, 3 & 19 and so on inside Logic.

So I went through a few quick trouble shooting steps:

  1. I looked at the direct output routing on the M7CL. Everything was patched one to one just as it should be.  Direct Out 1 was patched to Card Slot 2, Output 1-  Direct Out 2 was patched to Card Slot 2 Output 2. (the Dante Cards were in slots 2 & 3) That was all good.
  2. I looked at the matrix in Dante Controller on the Mac. Again everything was patched beautifully.
  3. I looked at the patching in Logic. 1 to 1, 2 to 2.

In theory, everything should work beautifully. It was time to dig deeper. We fired up Dante Controller on the Mac and took a look at the device info and network status.  This is where things got crazy.

Both of the Dante-MY16-AUD showed up in the device list, but only one had an IP address.  So we unplugged the cards one at a time from the network and each one showed up just fine. Then we plugged the second card in. The network assigned the second card the same IP address. That *might* explain the duplicated audio.

We took a quick look at Yamaha's and Audinate's (Dante's parent company) websites to see what the current firmware versions for the Dante Cards, and software.  We were a few versions behind. So we went through the process of updating Dante Virtual Soundcard the production machine, and my laptop, Dante Controller, and the cards in the M7.

We connected everything back up- aaaannnnd....(insert drum roll here) problem not solved. We were still getting duplicate data. If we manually assigned IP address one card wouldn't show up.  It was 3:30 in the afternoon. I had been at the church since 9:30. I had exhausted all of my options except one. Update the firmware on the M7CL. Unfortunately the church had a big production coming up and didn't want to do that and risk losing all of their scene data. I had to concede defeat and return home.

~Andy

 

Recording From Your Digital Console: Yamaha and Dante- Pt. 1 Configuring the Console

http://www.flickr.com/photos/gabitobalderas/6385082857/

Link to Gabriel's photo stream

The Rig

Now it's time to figure out how to do all of this with a Yamaha digital console, and a Dante network.

The computer of choice will be a 2010 model 13" MacBook Pro running OS X 10.7.5 "Lion." All of the Audio will be pumped into Pro Tools 10.

Our console today will be a Yamaha LS9-32 with two Dante MY 16-AUD Dante Network Cards. These are 16 channel network audio cards. They'll allow us to send 32 audio channels to our computer via CAT5 cable.  We're also going to need a Gigabit Ethernet switch. This will allow us to connect the two Dante cards to the audio network, then use Dante Virtual Soundcard on our computer. We wouldn't have to do quite as much work if we were using one of the newer CL series consoles with a Rio stagebox.

There's a few things to note here-

  1. I'm not going to go into every single detail- for eample installing the cards into the console is pretty easy. The guides on the Yamaha website cover that. I just want to touch a few things that might get missed along the way.
  2. This process might work on the first try for you. Or it might not.  I've had clients that haven't had any issues with a Dante set up. Then I've seen Dante networks collapse after the gigabit switch is power cycled- meaning they work one day, and they don't the other.
  3. You can probably tell by my last two notes that I'm not a huge fan of this setup.

Helpful Links

Here's a few helpful links for more information before you get started.  The first is from Yamaha and contains a few user and setup guides. You'll need to click on the tab titled "self-training." Read through these a few times. They're pretty helpful.

Yamaha / Dante MY 16-AUD Dante Network Cards

You're also going to want to make sure you have the latest software  and firmware updates:

Firmware, Software & Drivers

And you'll need a gigabit switch. As far as I know the Dante networks are a little particular about what hardware you use. There are some guides in the links below to choosing a switch.

Gigabit Switches For CL Series Consoles 

Selecting Network Switches

Getting Started

Installing the Dante cards is covered on page 12 of the current guide, available from the first link above. You'll need to do that, but in short they pretty much just plug into the back of the console.

So, once the cards are installed you'll need to decide what your clock source is. There is a lot of detail about that on pages 28-33 of the Dante-MY16-AUD User Guide.

Then you'll need to configure your direct outs. To do that you'll go to the patch editor on the console. Select the Direct Out patch tab. Then set Input Chanel 1 to Slot 1 Output 1. Set Input Channel 2 to Slot 1 Output 2, and so on.  When you get to Input 17, just set it to Slot 2 Output 1. In short Channels 1-16 Direct Outs get routed through Slot 1 Outs 1-16. 17 -32 get routed to Slot 2 Outs 1-16.

Last Step- this is really important.  SAVE YOUR SCENE. You may also want to consider "safe-ing" your patching. This prevents the output patching from being altered with scene recalls. For all the technical stuff you can stop reading here.

An Apology for the Delay

Now I need to apologize for taking so long to write this part of the series! I think it was October 2013 since my last post. I have to be honest- after I started I got rather bored with it. Having to work out all the computer details and things like that isn't that fun. I'd rather be mixing. Setting up a Dante network can be a bit involved, and sometimes the Dante cards on-board the consoles just don't want to act quite right.  That being said- Yamaha/Audinate have released several updates since I started writing this series that address quite a few problems. It's also noteworthy that the newer Yamaha hardware that are running Dante natively such as the new CL series consoles (CL1,3,5 etc) seem to work extremely well, as far as I know.

2014: A Few Things I'm Excited About

image I pay my bills by working at SE Systems in Greensboro, NC. It's a pretty cool place to work, especially if you're an audio geek. Not only are we a live events production company, and a pro audio, recording and lighting sales company, but we also have an on-site class/presentation room. We are going to be utilizing that room a lot this spring! I'm excited about that.

Rational Acoustics Smaart Training:

On February 25-27th, 2014, Jamie Anderson from Rational Acoustics- makers of Smaart- audio and acoustic measurement software- will be at SE Systems teaching users how to utilize the software. You can register here: http://www.rationalacoustics.com/events/greensboro-nc/

Smaart is amazing software. In short it allows you to "see" the sound. You can look at the frequency response audio systems. You can look at the reverb time in a room. You can look at the phase correlation between two different audio sources. There's really a lot you can do.

Worship Technology Information & Education Series

This spring we're offering a series of classes for all of the volunteer audio folks out there. Sound techs, sound person, techie, sound guy, A/V tech, worship leader- whatever this person is called at the church- this is for the person that wants/needs to know a little bit more.  Here's the basic layout:

Feb 15:

Audio Mixing and Multitracking.

March 15:

Worship Band Monitor Mixing and Personal Monitor Mixer Techniques.

March 22:

Loudspeakers, Wedge Monitors and Open Architecture Signal Processors.

April 19:

Microphone Techniques.

I hope to have more details soon. In the mean time you can keep an eye on www.sesystems.com

~Andy

Fixing the Source

A few weeks ago I read a blog that a friend of mine linked to on his Facebook page called "Fix It At The Source."  It's a great read on mic-ing technique, and even gives a few pointers on getting guitars dialed in to sound great through a PA system.  Check it out here: Fix It At The Source Sometimes You Have to Fix the Source

I've had the opportunity to operate many different types of audio systems.  I've used anything from beat up all-in-one box mixers to the newest DigiCo and Yamaha digital consoles. (and a lot of stuff in between those extremes.)  I always try to do the best I can with what's in front of me.  Sometimes there's stuff that I just can't fix.

Now before I go further, I have to confess I am a bit of a gear snob.

-BUT-I like to think I'm practical about it- meaning that there are well meaning and justified intentions behind it. I don't have the absolute best-ever-cutting-edge-gear but I like to have reliable stuff that works, and sounds good.  I want to use two illustrations:

I. I'm planning a romantic dinner for my wife on Friday night. So I'm going leave work at the normal time, stop at Harris Teeter, Wegman's, Von's, Safeway, whatever grocery store I happen across -grab two family size cans of Chef Boyardee Spaghetti & Meatballs, rush home and pop those bad boys in the microwave, plop the grub down on some paper plates and dig in.

OR

What would happen if I take off a little early, and stop at Joe's Italian Market? I could grab some fresh pasta, maybe some sun dried tomatoes, some dried herbs. Oh, and we'll add some locally made Italian sausage to that too.  Then I go home and bring all of this together, cook it, plate it, and serve it at a candle lit table?

Which one is my wife going to remember longer? Which one is she going to gush to her friends about?

II. Something to think about.  The person that works on your car probably doesn't buy his tools at Harbor Freight, or Northern Tool. If he did, he probably couldn't get the alternator off your car without cracking a socket, breaking a ratchet handle, or bending a screwdriver.  This stuff is very cheap. It might fine for a few little projects around the house, but you're not doing commercial grade work.

The Point

Having good source material is as important to a good mix, as good ingredients are to a good meal.  Having good tools that provide consistent results are also important.  They take the guesswork out and allow you to get to building good mixes.

A Fender Squire starter guitar kit is great for the beginning guitar player. A Casio keyboard is fine for practicing at home or learning scales. But there comes a point where you should strongly consider better sounding gear. It might even inspire you to play better!

Recording From Your Digital Console: DiGiCo and MADI- Pt. 2 Configuring the Computer

ConfigureComputer First Thing's First

I'm going to be bold here and hope you already have a basic understanding on how your computer works.  We'll assume that you've already installed Pro Tools.  I'm writing this while walking through the process on a 2010 model 13" MacBook Pro. It's running OS X 10.7.5 (Lion), and Pro Tools 10.

On a side note, I usually run my Apple computers one generation behind on the operating system. The reason for this, is that there can be a lot of heartache when you're running things on the bleeding edge of technology. I prefer to let someone else find all the bugs. I actually upgraded this machine from Snow Leopard to Lion shortly after Mountain Lion came out.  One huge reason for that is that Pro Tools wasn't supporting Mountain Lion at the time.

I am also not planning on upgrading to Pro Tools 11 any time soon. There's a lot of issues with plug in compatibility and things like that. If you want to read more about that then check it out at Avid's website here: Pro Tools 11 FAQ

What's Next?

First we need to unpack the UB MADI and install the drivers. The drivers are on a USB flash drive that's in the packaging under the UB MADI. Insert the flash drive, and open it up. Double click on the "Install DiGiCo UB MADI on Macintosh.pkg" file and the installer should start. Then just walk through the process. Then we'll need to set up the I/O in Pro Tools.

To do this, we'll start a new session. Start Pro Tools. I still have the start-up screen on my particular set up, so I'll select create blanks session. For audio file I'm choosing to run 24 bit, 96 kHz broadcast wave files (bwf.)

* Note that the UB MADI will set itself to whatever the audio input sample and bit rate is. If your DigiCo console is running 24 bit 48 kHz then you'll need to set your Pro Tools session accordingly or things won't play nicely together.

Once Pro Tools opens up, we need to change the playback engine. Click on "Setup", select "Playback Engine,"  go to the drop down menu and select "Pro Tools Aggregate I/O." A warning should pop up stating that "Selecting this playback engine will automatically save and close your session....". We are pretty sure we want to proceed so just click "Yes."

The session should restart. After things come back online we can set up our "Pro Tools Aggregate I/O."  That will be a post unto itself. Right now my plan is to touch on setting up a Yamaha console with a Dante network

Messing with Time and Space- A review of the Radial Phazer Bank

Radial Phazer Bank This blog will deal with time, and phase quite a bit.  Unfortunately I don't have the time to go into a lot of the details and physics behind all of it. There is already a lot of information out there. The Yamaha Sound Reinforcement Handbook is a great place to start.  You may also want to look at almost anything Dave Rat has written. So with that all in mind...(cue the drum roll)

The Radial Phazer Bank

Let me start by saying the Radial Phazer Bank is an amazing device. Phil at Radial Engineering sent one of these to us at SE Systems a few weeks ago. It's been sitting on my desk for a while because my studio was packed up for a move. The unit has garnered a lot of attention.  It's well built, by real Canadians.   It's got lots of knobs. It would look great parked in your rig. I have gotten lots of questions about it.

Note that this is not a phaser in the sense of it being a modulation effect, or "rotating speaker" effect, or a funky filter. It's designed to shift audio waveforms so that they sum and work together rather than pulling against each other and canceling stuff out.  Radial explains it well here: http://radialeng.com/r2011/phazer.php

The Set Up-

Ideally, I would have loved to test this device in my recording rig at home. I've got a Radial JDX amplifier DI, and some great mics for mic'ing guitar amps. I would have put a  Sennheiser e906 on the guitar cab. and plugged the JDX between the amp's speaker output and the speaker cab. Then used the Phazer on the JDX to align the signals. The only problem is, my studio still isn't unpacked.

This past Sunday I used the Phazer Bank in a live situation. The church I volunteer at uses two PreSonus StudioLive 16.4.2 consoles linked together to give us 32 input channels. We utilize the direct outs, and a few auxiliary sends to feed an Aviom personal monitor system. It's a decent console but has a few limitations.  We had the following inputs:

  • Lead Vocal
  • Backing Vocal
  • Acoustic Guitar 1
  • Acoustic Guitar 2
  • Bass
  • Keys (Piano Etc)
  • Synth
  • Electric Guitar 1
  • Electric Guitar 2
  • Room Mic 1
  • Room Mic 2
  • Kick Mic 1
  • Snare
  • Hi-Hat
  • Tom 1
  • Kick Mic 2 (Normally Tom 2)
  • Tom 3
  • Ride
  • Overhead (Not Used in Main Mix)
  • Drum FX
  • Vocal FX
  • Video Left
  • Video Right
  • Wireless Hand Held (For announcemnts)
  • Wireless Headset (Pastor)

You will probably notice that I have two kick drum mics.  I used one dynamic mic inside near the beater to pick up the attack, and a condenser in the bottom of the to pick up the "boom."  In a perfect world I would have put a slight delay on the kick drum inputs so that the kit would be "time aligned" with PA system.

Having the kick and PA "aligned" would mean that the speakers are delayed until the sound from the kick travels to the point in space where the speakers are, then the speakers fire.  Everything is moving in the same direction at the same time. It can make the difference between hearing the kick and feeling the kick.

Plugging It In

So the PreSonus StudioLive console doesn't have input delays. But I can at least phase align the kick with the PA system. This would mean that the peaks in the drum sounds are lining up with the peaks of the drum sounds coming out of the PA, and the troughs are lined up with the troughs.  I inserted one channel of the Phazer Bank each of the two Kick Mics. (Using the inserts on the appropriate channels on the console.)

I left the condenser mic muted while adjusting the Phazer on the dynamic mic. I swept through the shift settings until it sounded full.  Then I brought condenser into the mix, swept through the shift setting on it until it sounded super full. The results were amazing. I could feel the kick in my chest and the band wasn't even very loud. In fact before the service, the pastor asked, "why does it seem so loud? You're only hitting 87 dB SPL on your meter." Needless to say I was immediately impressed by the Phazer Bank.  I want to buy one. Now. It sounded great!

But Here's the Rub

The Phazer Bank works extremely well. Unfortunately on this particular audio console the channel inserts are pre- direct out.  Our direct outs feed the personal monitor system. This means that while I was pulling the kick into alignment with the PA, I was pulling it out of alignment with the drummer's in ear monitor mix. So he was losing the "kick" and "punch" that he usually had. There are ways to fix that, but it's going to take a few more pieces of gear... Or I can just revert to making "good" sound rather than "great" sound. But I still want a Phazer Bank.

~Andy

Recording From Your Digital Console: DigiCo and MADI- Pt. 1 Configuring the Console

SD9andUBMADIThe Rig For this particular example we're going to use a DigiCo SD9, with two D-Racks, and the digital snakes. This is probably my favorite set-up. It's super easy to use, and quick.

To interface the SD9 to the computer, we're going to use a DigiCo UB MADI- MADI to USB 2.0 interface. The computer of choice will be a 2010 model 13" MacBook Pro running OS X 10.7.5 "Lion." All of the Audio will be pumped into Pro Tools 10.

Configuring the Console

On the hardware side of things, you'll need a 75 ohm video grade cable terminated with BNC connectors to connect the console's MADI output to the UB MADI.

Configuring the console has to be one of the easiest set-ups I've ever seen. We are going to copy the inputs from the D-Rack to the MADI output of the console. This taps into the input channels just after the pre-amps, before any EQ or other processing. Here are the steps:

  1. Press the "Master Screen" button on the console.
  2. Select the "Setup" tab on the top right corner of the screen.
  3. Select "Audio I/O."
  4. Find "Rack 1" on the left column of the open window and select it.
  5. Find the "Copy Audio To" button on the top right area of the open window.
  6. Select "3: MADI" in the drop down menu.

Boom! You're done! If you're using 2 D-Racks you would just repeat the process to route the 2nd rack. We can now transmit our audio to the computer. Stay tuned for Part 2, when we will install the UB MADI software and configure Pro Tools 10 to receive the incoming audio!

Recording From Your Digital Console: Choosing Recording Software

Picking Up the Lingo I'm going to start off and just throw a few terms that I might use in the series out there. That way we're all on the same page, and I don't have to type the words "recording software" every other sentence. This may also help as you explore the interwebs and research what options might be best for you. So, without further ado-

The Terms:

DAW- This stands for Digital Audio Workstation. This is what we generally call the software, whether it's Pro Tools, Cubase, Studio One, Reaper etc.

ITB- This is simply an abbreviation of IThe Box. The box in this case is your computer. Some recording engineers prefer to mix out of the box, meaning they're using an audio console to mix their recording projects. Some prefer to mix in the box using the faders in the software.

Plug-Ins- Plug-Ins are virtual equalizers, compressors, reverb and other effect units.  In analog world we would typically patch or plug these into our mixer using cables. Many times ITB we just use a drop down menu.

DAW's- The Contenders:

I'm going to start by answering this question with a question. Which DAW do you like? There are free options like Audacity (which doesn't play nicely or at all with Audinate's Dante Virtual Soundcard. It could have been a problem on my end.) Reaper is a nearly free option ($60 for students or non-profits $255 for everybody else.) Both of these are distributed directly from their websites

Then there's the paid options.  Most people have heard of Pro Tools. It's an industry standard in the professional recording world. Then there's others like PreSonus' Studio One, Steinberg's Nuendo & Cubase  family of products, Sonar by Cakewalk, and Apple's Logic. Each one of these has it's own set of advantages and disadvantages.

So Which One Do I Choose?

It really depends on your end result, and your workflow. Personally I don't have a lot of hands-on time with the Steinberg family products, or Sonar. My two personal favorites are Pro Tools and Studio One. I typically use Pro Tools the most.  I'm just used to the workflow, the keyboard short cuts, and I like the routing matrix.  I would encourage anyone with a little time on their hands to download demo versions of any of  these software packages and try them out. See which one you like.

~Andy

Recording From Your Digital Console: Choosing A Computer

MacVPCCutting to the Chase Buy a Mac.

But Seriously

Buy a Mac. (Are we beginning to see a bias?) This article is a bit tongue-in-cheek.

My Case Against Windows

Have you ever been computer shopping? On the Windows side of things you have Dell, HP, Gateway, Acer and others. Then there's the specialized machines from companies like ADK, Alienware, or Music XPC.  That's seven different manufacturers that I've listed off the top of my head.  Some of those companies have as many 15 different product lines. Each one of these uses different chipsets, different USB and Firewire buses. If you want to see what's available feel free to go to a website like Tiger Direct (www.tigerdirect.com) or New Egg (www.newegg.com.)

My point is, the operating system, Windows, has to be compatible with all of these different machines. It also has to work with all of these different parts. That's a lot of programming code.  There's a great opportunity for something to just not work quite right. When things do go wrong, who do you call? Microsoft? Dell? The mother board manufacturer?

Finally, what version of Windows do you buy? Home, Professional, Ultimate? 7 or 8? Lots of choices. These choices can affect how your computer and audio hardware interface with each other. I will say this. If you are considering Windows 8 for a recording/production machine- wait. The various audio software/hardware manufacturers still need time to update software/hardware drivers. (I would actually say the same thing if Apple just released a new operating system.)

The Argument for Mac

There was a period in my life that I worked in the Keyboard & Recording department at a chain music store. Typically if a customer purchased any recording, or music creation software from me, I'd offer to help them get it installed if they had issues. Granted at this particular time there was a version of Windows called Media Center Edition. That particular version would absolutely not work with external sound cards. Period. Ever. Other than that, it would often take multiple install attempts to get a particular software working. I rarely had Mac users come in with trouble. Things boil down to this. How much time to you want to spend trouble shooting your gear, vs how much time you want to be using it.

If you happen to visit the Apple website, you'll notice there are only five series of computers.  That's considerably less than the plethora of  Windows options. Your choices are two laptop lines, and three desktop lines. Then you have to take into consideration that Apple builds it's own computers, and their operating system (OS X.) That has to guarantee a certain level of cooperation between the software and hardware.

My final argument is that OS X, Apple's operating systems supports Aggregate Audio Devices.

What Do Aggregate Devices Do for Me?

In simple terms they allow you to use multiple sound cards within OS X or within applications that support it. Why is this useful? Well, let's look at our scenario from the previous post in this series. In this case there was a digital audio console at front of house, pumping 32 channels of audio to a computer back stage via a Dante network.  The computer was using Dante's Virtual Soundcard. Unfortunately, because it was a Windows based machine the only audio device it could use was the virtual sound card.

This was extremely problematic for using local audio monitors. One work around would be to close the session, and then re-open it.  Then an Avid M-Box, PreSonus Audio box, or similar device could then be used to connect studio monitors. This would not offer real-time monitoring of input. A second option would be to add another Dante device to the network, in the broadcast room to connect a pair of studio monitors to.

By setting up an aggregate device inside Pro Tools, you can then use the Dante Virtual Soundcard (in this particular case) for input, and select an M-Box or other small interface to use as output for local monitoring.

~Andy

Recording From Your Digital Console: A New Series

digitalrecorder Introducing a New Series

I've decided to take a short break from the lighting world to focus on two other parts of my field: live sound and recording. I may have mentioned before that I work at SE Systems in Greensboro, NC.  SE Systems is a pro audio, lighting and video sales and production company.  A month or so ago we had a customer that was setting up a recording system for a house of worship. He was having some issues getting things working together well so he gave us a call.  I ended up walking him through the setup over the phone. I have since decided to write about it so that others may benefit from what we learned figuring out this job.

The Set Up

All of the church's stage inputs were sent from the stage to the console via an analog snake. The console, a Yamaha LS9-32, had two Audinate MY-16-AUD Dante Network Cards installed. The console's direct outs were routed through the Dante cards. The Dante cards fed 32 channels of audio, via Cat 5 network cable,  to a custom built computer in a broadcast room backstage. The computer used Dante's Dante Virtual Sound Card to interface with Pro Tools 10.

The Problem

Dante Virtual Soundcard turns the computer's network card into an audio interface, allowing the computer to bring in audio from devices using the Dante format. The church had an Avid M-Box to connect to the computer for connecting a pair of studio monitors in the broadcast room.  Unfortunately with some computers and some recording software, you can only use one sound card or interface at a time. This meant that there was no local audio monitoring for the recording engineer to listen to, unless he saved the recording session, and re-opened it using a different audio interface on the computer. Another option would have been to get a Dante device for the broadcast room, but that would have been rather expensive.

The Solution

For now, this particular house of worship is simply just recording their services and saving the session. Then mixing the recording down later using the Avid M-Box for monitoring. It works but, I think it could be better. I'll unpack how to do this well over the next few post.  My plan right now is to break this down into a few sections:

  1. Computer Selection- What to look for in a recording computer
  2. Digital Audio Workstation Selection- This is the actual recording software. We'll reffer to it as the DAW or DAW softare just to save space from here on out.
  3. I'm running into two digital audio formats pretty often at work. A lot of people use Yamaha consoles, and the Dante cards are widely available. DigiCo is also making huge in-roads into the industry. They use a digital audio format called MADI.  I'll take some time and break down how to set up each type.
  4. Finally I'll go over some tips on how to set up your DAW, how to patch things, and some cool tricks we can use to solve monitoring problems.

I'm going to wrap this up by saying that these aren't the end-all/be-all solutions. They are tips to get started fast. I will also warn you that I am extremely opinionated. I've been helping people get recording systems set up for many years. I've run into all kinds of problems. There are some systems and some DAW's that I have run into problems with. There are other systems that I have run into fewer to no problems. I have a heavy bias toward the latter. I also prefer certain things just because they fit my particular workflow or I like the way they look or feel.

~Andy

Out In the Field: When Things Go Wrong...

View From The Desk

The Backstory:

One of the other hats that I wear is volunteering on the production teams at my church, Salem Chapel in Winston-Salem, NC. Typically, once a month, I serve as the audio engineer for three services over one weekend.

Salem Chapel is a mobile church.  In many aspects it is like working for a small production company.  Every Saturday we show up at a local middle school at 2:00 PM, and unload a trailer full of equipment.  Teams of people work around the school setting up different areas of the church. My primary job is to unload the road case that contains the audio consoles, and plug it all in, and have it ready for the worship team to practice at 3:00. Our first weekend service kicks off at 5:00 PM. Things wrap up at 6:30 and the production team and church staff are usually leaving around 7:00 after debriefing.

Sunday morning the production and worship teams arrive at 8:00 for a short practice. The first Sunday service runs from 9:00 to 10:30, and our final weekend service runs from 11:00 to 12:30. Then we pack everything back up, and load it back onto the trailer.

The Problem(s):

Our equipment rides in the back of a 16 foot trailer, bouncing up and down with every bump in the road.  Things shift in road cases. Sometimes gear fails. Then add the fact that we get the get the equipment out and have to connect audio, lighting and video connections,test the system and have it ready to roll for practice in about an hours time. Sometimes we just forget to plug stuff in. Whi is where our first problem this weekend entered.

 "I Can Hear You, But I Can't Hear the Rest of the Band."

This is not something I really want to hear anyone in the band say. It means there's a breakdown somewhere in the signal path. There was a very valid reason that my friend Jake couldn't hear the rest of the band. I could look at the meters on the console and tell I had input on all of the channels. I looked at the lights on the front of our Aviom Personal Monitor input module. It was powered on. I also knew that the Aviom system was on, because Jake could hear me. Then I remembered, during set up, I had not plugged in the DB25 connectors that feed the monitor system. The reason that 3 vocals, 2 acoustic guitars, 2 electric guitars, an electric bass, and two channels of keyboards could not be heard was that their direct outs were not connected.

The DB25 Connectors I forgot to plug in...

Fortunately, it wasn't a huge deal. We caught the problem early, really even before practice started. What can we learn from this?

  1. At some point we might want to consider making a checklist or punch-sheet for setting up the system. That way we set things up quickly and consistently every single time.
  2. To quote one of my college instructors, Thomas Johnson, "It's all about signal flow."  It is absolutely critical to understand how signal flows through your PA system. Where does that sound come from, and where is it supposed to go? If I can hear this, and I am supposed to hear this and that, and I cannot hear that, then why can't I hear that?

Take the time. Learn your system. Ask questions. Read manuals. Even take the time to look at the block diagrams in the manuals. It can save you from having a really bad day, or at least save you a bit of stress.

~Andy

Coming Up...

Found this on Flickr, pic by Cordey: http://www.flickr.com/photos/flygraphix/3244828717/ I just wanted to take a moment and throw out a quick teaser of what's coming up.

Before I continue with "Programming Inexpensive Controllers", I'm going to revisit the  "Illuminating DMX" series. I want to take some time to go over DMX protocol charts, which are found in the manuals of most lighting instruments. In short these charts explain what parameters of the instrument are controlled by what DMX channel. I feel like it's important to understand this, so that you can develop a strategy for addressing your instruments, as well as programming scenes.

I would also like to take a trip through audio world for a little while.  Some of this will probably sound like a foreign language to some of you, but that's ok.  I'll try to unpack this stuff in depth sometime.  I recently had a customer purchase two Dante-MY-16-AUD digital network cards. These cards allow the user to take 16 channels of audio, bi-directionally into and out of a console via Gigabit Ethernet. They were installed into a Yamaha LS9 digital audio console. The goal was to take 32 direct outs from the console at front of house, and feed a computer in a broadcast room in another part of the church, then record that using Pro Tools 10. I'll take some time to explain how we had to patch the direct outs in the console, set up the Dante Virtual Soundcard on the computer, and build an aggregate audio device in Pro Tools so we could have local monitoring in the broadcast room.

~Andy

Out In the Field: Console Tape & Sharpies

by Andy Barnhill

Console Tape and Sharpies

Console Tape and Sharpies

Two things every audio engineer should have with them at all times are console tape and a Sharpie. This can be super handy for the volunteer audio engineer in a house of worship.  Why console tape? There's a few reasons:

  1. It's whiter than masking tape, so there is better contrast between the black marker and the white tape. This is especially helpful in low-light situations. Blue painters tape can work in a pinch, but the contrast between the tape and the ink is terrible. Note that I emphasized "can" and "in a pinch." Blue tape isn't the best option.
  2. It (typically) doesn't leave residue on the console, or remove the paint. I say typically, because if you do leave the tape on the mixer for a long period of time, it can gum up a little. I will also add that one Sunday, the tape did pull some of the gray paint off of the console in the picture above.
  3. It's a really great way to develop rapport with the worship team. You might be asking yourself, "how is this so?"

I typically use 3/4" tape, often two strips just below the faders. On our console, we already have a strip of tape above the faders that label the inputs with things such as "Lead Vox," BG Vox 1," BG Vox 2," "AG 1," (For Acoustic Guitar) and other inputs.

What I like to do is put the persons name below the fader. This can be especially helpful when there are new members in the worship team, or even different members each week. It gives me a reminder of the person's name at each position in the band.  I don't have to think about it, I can just look down.  I could ask, "Hey, backing vocalist, would you mind singing a little louder?" On the other hand, if I have the person's name under their input, I can make a more personal connection, for example, "Jake, would you mind turning the output on your acoustic up a little bit?"

It keeps me from having to stop and ask something, like, "Hey, I'm sorry, what's your name? Oh yeah, well could you ...." Any time you can connect with the worship team on a personal level, it helps create unity between the band and the technical team.

Think about it like this: are you more prone to respond positively to someone asking you to do something by name, or if someone asks," Hey, sound guy, umm can you turn up my monitors?"

Documentation Is Key

by Andy Barnhill

As a technician, designer, or engineer documentation is absolutely critical. Missing information can bite us in the future. Detailed information can be extremely helpful.  Recently Peter Gabriel released a 25 anniversary edition of the album So.  In an excerpt from Peter Gabriel's website, Richard Chappell shared the following:

"Ian made some experimental changes but both he and Peter agreed that it sounded great as it was. Incredibly Ian actually had the original notes from the session 25 years before! They ended up approaching the mastering in the same way that they had at the very first session."

For more about Peter Gabriel's So 25th Anniversary, check it out here: http://petergabriel.com/so25/