My Parallel Processed Bass Rig Part 2: The Gear

This is the second part of a series I’m writing on parallel processing bass guitars. In part one I discussed some of the benefits of parallel processing. You can check that out here. In part two, I’ll talk a little bit about the gear that makes up my particular rig. In part three I’ll get to the signal path. Before we get into the signal path for my particular setup, I’ll tell you a little bit about what’s in my rig.

Here is what’s there:

How/Why I Chose It:

P-Bass

I purchased the P-Bass almost 10 years ago because I needed a bass and a friend of mine was selling it for $250.00, with a hard shell case. It was a deal too good to pass up. It’s a three color sunburst finished, four string with rosewood(I think?) fingerboard. I have no idea what year it was made.

About a year after I purchased it, somehow the pickup died in it, so I replaced it with a Fender Original P-Bass Pickup. I also installed a stacked volume/tone pot, and put a rotary switch with a “chickenhead knob” in the former tone pot location. The switch selects between four options.

  • Mute (Kills the output of the bass)

  • Volume & Tone Bypass (sends signal straight from the pickup to the output jack)

  • Uses a .047uF capacitor on the tone circuit (pretty standard for P-Basses)

  • Uses a .047uF vintage style capacitor on the tone circuit. (I’m planning on replacing this because it makes absolutely no difference to the tone of the bass. Don’t worry it wasn’t one of those $40.00 ones. I’ll probably go with something that will cut highs at a lower frequency than the .047uF. So don’t subscribe to the hype around “designer” or “vintage reissue” or whatever capacitors)

Recently I re-filed the nut and tuned the bass B, E, A, D. I play at church and we end up playing in the key of B, C, C#, and D fairly often. I wanted to have an option of being able to walk a bass line down instead of having to go higher on the neck as often. This lets me do that, and it’s cheaper than buying a five string bass. I will probably end up with another bass at some point because there’s been a few things I’ve wanted to play on my own that really need a G string.

VT Bass DI

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The first Tech 21 VT Bass DI I picked up because I love the sound of an SVT Classic and 8X10 Cabinet. I do not want to move either of those, and I wanted something at home to practice with. The great thing about the VT Bass DI is the character knob allows you to sweep from classic Ampeg SVT tones to the flip-top amp, to more modern tones. So it’s extremely versatile. Then I decided I might need to switch between gritty clank SVT and smoother “woodier” sounding flip-top tones live, so I just got another one. Plus it never hurts to have a spare.

WTDI

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From about 2000 to 2010 I worked in several music stores. One of the stores I worked at was an Eden Amplification dealer, probably during Eden's golden era. We're talking back when the WT-800 was the bass amp to have. I worked with a guy named Scott, who is an amazing bass player, and he loved the Eden stuff. In fact in his personal rig, if I remember correctly he had a David Series 410 and 212 cabinet. Not common but it fit his style well. The thing that stuck out to me is that it just sounded so full. If the WTDI got even a little close to that, to me, it’s worth it.

Woolly Mammoth

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The Woolly Mammoth was an impulse purchase. I usually do lots of research, read some reviews and go to stores and try gear out before I buy it. That was not the case with this Woolly Mammoth. I used to follow Zvex and a bunch of other pedal manufaturers on Instagram. Zachary Vex posted that they were selling some pedals off of their workbench/repair area. Some were prototypes, some were modified and experimented on etc. They posted a picture of a Vexter Woolly Mammoth. I decided I wanted it. I contacted them and purchased it. At first glance it’s not particularly remarkable. It does have a piece of green tape on it that says “Sockets” on it. The transistors are mounted in sockets, SO if I wanted to, I could try out different ones. But I probably never will. I also do not regret buying this pedal. It sounds super awesome. It is way too fuzzy to use in my normal church gigs. Very Chris Wolstenholme/Muse type fuzziness. But it lives on my pedal board. And I use it. Because I can.

ES-8

This is the brains of my rig, and really allows me to be flexible, and have different bass tones available for different songs. If you’re not familiar with effect switching systems, or pedal switching systems- they allow you to set up your individual guitar or bass effect pedals, and then with the press of a single button engage one or more pedals at the same time. I originally purchased this to use with my guitar rig, but ended up needing it for bass too. So for now depending on what gig I’m playing I switch it out a lot- but I’m also working on getting that resolved as well. Here’s why I chose the ES-8

  • It’s got 8 loops (or slots) for pedals

  • Loop 7 can be a mono out and stereo return

  • Loop 8 can be stereo send AND return

  • You can change the order of pedals per each preset if you so choose

  • It has MIDI in and out

  • You can set it up to split your signal and run parallel, with up to two splits

  • It has two outputs

  • Has a volume pedal loop (that doesn’t count against your 8)

  • Has a tuner output, so you can mute everything and tune

  • It has a computer based editor so you don’t have to navigate with the tiny buttons and on-screen menus on the unit itself

So that’s the basic rundown. Come back soon to learn about the signal path. If you haven’t yet, be sure to check out part one of this series “My Parallel Processed Bass Rig Part 1: Why Parallel Processing?

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.

Projection Mapping for Worship Pt. 2: The Software

I'm going to talk more about the software that we used for the "video wall" for Calvary Church's Immerse Night of Worship last February.  In my previous writing, I talked a little bit about the design and overall signal path. You can check that out here: Projection Mapping for Worship Pt. 1: The Design

We used two pieces of software to achieve our final result. One thing to note, is that all of the software I'm talking about here is Apple OS X based. I apologize in advance to any Windows users.

The first was VDMX by Vidvox. VDMX handled all of our media server duties, effects generation, and masking. The second software was called Madmapper by Garagecube/1024 Architecture. Madmapper took care of all of our geometry corrections. Think of it like keystone correction on steroids. VDMX's output was fed into Madmapper via a protocol called Syphon.

VDMX

VDMX 5 by Vidvox, Screenshot of interface for the Immerse event at Calvary Church in Greensboro, NC from February 28th 2016.

I started tinkering with VDMX several years ago after hearing about it from my good friend Josh Orton. The church he was attending at the time was using it to feed content into a projector aimed at their upstage wall.  They would mask out areas on the stage they didn't want to show video on, or use it to highlight specific things during their worship sets. I was searching the internet for more information about it one day and found out one of my favorite musicians, Tycho uses it as well. I became pretty smitten with it at that point.

Vidvox's VDMX software is a Mac-based media/video server software.   In fact, their  tagline is "Professional VJ Software."  Think of a VJ as a DJ, but rather than using records, discs, or now audio files, a VJ would use video files. In my humble opinion, calling VDMX VJ software is an understatement. It would be like calling Mount Rushmore a statue, or the Grand Canyon a riverbed.

Think of VDMX as a non-linear video playback and manipulation system. You can load video clips into it much like a video sampler, add many different effects such as blurs, color swaps, flips, cropping, masking, speed changes, reverses, kaleidoscopes, all sorts of effects and plug-ins.  You can almost do anything that you would do in a video editor but in real time.  If you're familiar with Ableton's Live software for audio, in the same way, that you can work in real-time, triggering sound clips and samples, you can do the same thing with VDMX. Live performance is what VDMX does well. 

Most of VDMX's functions like video clip triggers, cross faders, speed control parameters, screen placement can all be assigned to either keys on your Mac's keyboard, a MIDI Controller, or OSC based controller.   There are even plug-ins to enable the use of Nintendo Wiimotes. All of these options present the user with an infinite sea of ways to control their video rig.

VDMX is a remarkably stable software. It's designed to process and render all of its video content on your Mac's graphics processing unit.  The minimum system requirements are a Mac computer with an Intel Processor (pretty much every Mac made at this point,) running OS X 10.10 or later.  Recommended System Requirements include a Nvidia or ATI graphics card and 4+ GB of RAM.


I am about share something that I do not recommend doing.  VDMX 5 will run on my Mid 2010 13" MacBook Pro, 2.4 GHz Intel Core Duo machine with 4 GB RAM.  That machine does have a Nvidia GeForce 320M 256MB graphics card in it.  When I purchased this machine, the 15" MacBook Pros had dual GPUs on board, and would switch to the more powerful card when running video or graphics editing programs. My MacBook Pro has an underpowered GPU for this sort of work.  BUT VDMX runs on it. I've never pushed it super hard or loaded a lot of effects, but I can usually work on an interface setup or get a proof of concept working on it just fine. I just wanted to share this to give you an idea of how stable it is.~ Andy


Madmapper

Screenshot of Madmapper.  The pattern on the left is the input from VDMX, and the Lincoln Memorial is the output side.

Madmapper is software that allows the user to do several things to the output of a projector. The first is geometry adjustments- In a perfect world, we would always have our projectors pointed directly and squarely at the surface that we're trying to project on.  In some cases, that can't happen so we can use Madmapper to correct for that. For example: Trying to project an image onto two adjoining sides of a cube.  Two parts of an image can be picked up from the input side of Madmapper, then "wrapped" around the corner of the cube so that the image is nice and square from the viewers perspective.  For the Immerse project, we had to mount our projectors higher above the projection surface than they were designed to be.  We also didn't have the required space on stage to center the projectors in front of each wall. So we warped the image in Madmapper to stretch it back into the correct alignment.  Sadly I didn't take any screenshots of Madmapper from this project, so the Lego Lincoln Memorial above will have to do.

~Andy

Moving DMX onto a Network: the Elation Professional EZ Kling

Where it all begins...

I've already written a few blogs about DMX-512 (commonly just referred to as DMX.) One limitation of DMX is that generally you can only control up to 512 channels of things that need to be controlled. Those 512 channels are called a universe. When we were using PAR cans 512 channels was a lot of lights. Now, with moving heads and LED units using many channels each- it's easy to burn up a universe quickly.  For this reason, some lighting consoles have multiple outputs for universes, four or eight or so.  This means that you end up running four times, eight times, whatever times the cable to the stage- So we really need a better way to do this...

Enter Art-Net

Art-Net is that better way.  I'm not saying it's the best, or only way, but it beats running multiple DMX cables to the stage.  Short version- some very brilliant people at a company called Artistic Licence in the UK figured out how to take all of the little bits and bytes in the DMX-512 protocol, stuff it into little packets and cram it all through modern ethernet cables. In fact we can now transmit over 30,000 universes over a gigabit network.  So let's math that out-

Art-Net III can carry up to 32,768 universes, each has 512 channels of control in it for a whopping 16,777,216 channels of control.

IF we were to use that to control RGB LED lights, that would give us just over 5 million pixels. Not that I'm advocating buying 5 million lights- but if you're considering it, please, please call me! I'd love to help you out with that!

SO- It's not a new protocol- just a new way to transmit it. They've also been super generous and released it through open source and made it easily available so that other people can use it.  I'm not going to go into a lot of the technicals on how it works, there's a great article on Wikipedia here: Art-Net and a link to Artistic-Licence's web site here: Artistic Licence. All of this makes it easier to get into the world of pixel mapping. Simply put- this is taking an RGB LED lighting instrument and using as a part of a video, or some type of graphic display.

So Why The Sudden Interest?

Full disclosure- I currently work at SE Systems in Greensboro NC. We're a live production company, as well as a pro audio, lighting, recording and video sales company. I primarily sell lighting and recording equipment.  Part of my job duties include finding cool new things to carry.

One recent discovery is the Elation Professional EZ Kling. It's a simple little black box network device that will either-

a. Convert KlingNet (basically put- a proprietary video to lighting protocol for pixel mapping lighting instruments) and Art-Net to DMX, or

b. Convert DMX to KlingNet or Art-Net.

So How Can I Use This?

There are several different ways to use the EZ Kling.  Here are a few examples:

  • Expand the DMX output of certain lighting control software, or consoles.
  • Pixel map lights that don't have Kling-Net, or Art-Net natively.
  • Capitalize on existing network infrastructure to control lights in multiple areas, by combining with an Art-Net input device such as an Elation Professional eNode 4 or eNode 8.
  • You could also potentially control small numbers of lights using a Wi-Fi network (I really wouldn't recommend this for large numbers of lights, or super critical lighting work.)

Stay tuned over the next few months as I explore some tricks using these handy little devices.

~Andy

Lighting Control: How to Get More Bang for your Buck..

This is going to be a bit of a longer post, but there's a bit to cover.  A little while ago a person identifying himself  as  StevieWonder posted the following comment on one of my posts:

"Also wondered if you have any recommendations on controlling dmx with software and an ios device or a cheap netbook, or if a standard controller would do the trick? Any advice you can give would be much appreciated!"

This immediately raised one question in my mind. How does Stevie Wonder program lights?

The Dilemma

Until very recently, complex lighting control wasn't very affordable. Even super basic fader controllers to use with a basic dimming system were still around  $900 once you figure in sales tax for 24/48 channels.  They won't control movers well and still use microplex-  which is an archaic control protocol. (Although when looking at what ArtNet and RDM are bringing to the table DMX is archaic too.  All of this is a discussion for another day)

So then we take a look at the basic controllers like the Chauvet Obey Series, or the Elation/American DJ DMX Operator series.  They're great if you only have say 12 different fixture types, and are maybe recalling movement macros, or color macros on your lights. You're probably not going to program the Kiss reunion tour on one of these. I feel like it would be similar to hanging drywall with a screwdriver.  You can do it. But it's going to take a lot of effort.

The Solution

I'm about to say something really crazy. This is especially insane considering over a year ago I posted this blog about choosing a computer for recording- Recording from Your Digital Console: Choosing a Computer. The most cost-effective way to get a lot of features in a lighting control system for not very much money is to buy a touch screen tablet PC running Windows 8.

Many of the current lighting control software options offer remote control via iPad, iTouch, iPhones, and some Android devices, but you have to have a computer to host the main part of the program. Why not just skip a step and buy a touch screen tablet? It will be cheaper than buying a desktop/laptop machine and an iPad. There are some great deals on new and refurbished units out there. You just have to shop around. New Egg and Tiger Direct are two great places to start looking. (Unless we happen to start carrying tablets at work- at which point I'll edit this post and add a link.  Yes, I know shameless plug, but I gotta buy gear too. And eat.  Bacon ain't as cheap as it used to be.)

The Software

 

For lighting control software there are two main titles I would look at.  One is the Avolites Titan One dongle. The other is Chauvet's Show Xpress. Conveniently, we carry both of these at SE Systems. I'm only going to highlight a few features of each right now. I'll probably go a little more in depth later.

Avolites Titan One

Titan_One_Avolites_900

This. Is. My. New favorite thing! The Avolites Titan one dongle has quite a few advantages working for it.  The first is that it's the same software that all of the Avolites lighting desks use. The Titan One all the way up to their Sapphire Touch use the same exact software. This means if you learn how to use one, you can use them all. Here's a few other advantages:

  1. Price Point- the retail price of this package is only $250.00 USD. You have to be kidding me! Pro level lighting control for less than $300. (OK well you still have to have a computer but still)
  2. Cue Stacking- This is a great feature that allows you to take multiple cues, and assign them to one fader. For example, say you have a look programmed for your house lights, a look for your upstage (back wall) uplighting, and a look for your stage/front lighting. You might have a volunteer running the console, or you may just want to be able to touch one fader to have all three of those things change at one time.  You can do that.
  3. Avo's Quicksketch-  This feature lets you quickly assign a picture or scribble your own to assign to buttons and faders. So you can have a picture that means something to you. Like a guitar for a guitar solo look. Or a picture of Bob if Bob is speaking this week. Or a big red button for your big red look. Or a smiley face with a big hat when the band covers "Happy." PharrellHappyArtNews600
  4. Pallets, Shapes and Pixel Mapper effects-  These are tools that allow you to build shows quickly and efficiently- Think of pallets as collections of attributes.  I have a box of colors. I have a box of moving light positions and movement patterns. I have a box of chases and color changes that I can draw from.  Then I can take these attributes, select the ones I want and place them in a cue or cue stack.
  5. YouTube- There are a lot of YouTube training videos linked on Avolites' website, or you can just look for Avolites' YouTube channel.
  6. The Manual- There is a very detailed manual available for this software that goes over each of the features.

Chauvet's Show Express

showxpress7-l

I've probably spent more time using this particular software than the Avolites. Part of that is that the Avo software hasn't been available quite as long. (At least not at work.) Here's a few things I like about it:

  1. Live Mode: This basically gives you a panel of buttons that you can use to trigger scenes or looks. Very simple interface for a beginning user or volunteer operator. Live mode can also be remotely operated via an iPhone or Android based app. (But if you're running on a Windows 8 machine, there's no need to. Unless you want multiple pages of scene triggers- which might be super handy.)
  2. You can trigger multiple scenes in different windows of live mode. So if you have your uplighting blue, and your stage lighting green- as long as you have those scenes saved separately, you can recall them individually, but at the same time.
  3. Start in Live Mode:  You can actually set the software to start up in "Live" mode so that the operator doesn't have to deal with the set up or programming options.

So this is just a super basic overview of two systems. I may write a more detailed write up of each system in the future.

~Andy

Take a Look at This: DOME PROJECTIONS | Joanie Lemercier

Over the years I have come to realize I am really a fan of all technical arts. One area that fascinates me is projection and projection mapping. I've posted a excerpt and a link from Joanie Lemercier on a dome projection project below:  

The Satosphere is a 18 meters diameter dome on the roof of the Society for Arts and Technology [SAT] in Montreal. It opened in 2011, and has a 360° x 210° spherical screen, equipped with 8 video projectors, 157 speakers, and has a team of designers and technicians working in the space, doing content creation as well as active R&D on immersive technologies and contents.

via DOME PROJECTIONS | Joanie Lemercier.

 

~Andy

This Post is Unrelated: Building a Telecaster from Parts

Beginings... The electric guitar is really the beginning of my foray into the world of professional audio and lighting. Somewhere around 17 years ago I had an Ibanez electric guitar on layaway at a local music store.  Looking back it was probably one of the ugliest things I've ever owned. It had a nice blue metallic flake finish. Think: bass boat. I might have been manipulated into buying it because I had seen Joe Satriani in concert the night before. Joe played Ibanez guitars, so why not right?BlueMetalFlake Google is a wonderful invention.   I was able to find a picture of an old advertisement with such a beast featured in it. It's tacky, right?

One day while making a payment I asked if they happened to be hiring, and the rest is history. I worked there for many years, eventually becoming a part of their audio install team.

My current employer, SE Systems  is a full line pro audio, lighting, recording and video dealer. Unfortunately, (or fortunately depending on how you look at it) we don't sell musical instruments.  I've also been more involved with sound reinforcement and light over the last few year rather than actually playing guitar so my gear purchases have been more in that spectrum.

Until now...

About a year ago I decided I wanted a new or used guitar. I already had a Fender Hot Rodded American B-Bender Telecaster. What I really wanted was a Strat, but I really like the feel of a maple Telecaster neck. I decided to see if I could build a Strat with a Tele neck from parts.

'52 Reissue Body Mid-Strip

I mentioned this desire to a friend of mine and he said, "I have a Tele body I'll sell you for $20.00.) It wasn't what I was looking for, but for $20 I figured I'd buy it. As it turns out, it was a Fender '52 reissue body in a butterscotch blonde finish. I also bought an off-brand "Strat" style guitar to take apart and Frankenstein into something else. I spent some time scrounging on eBay for parts for the Strat project. Meanwhile Telecaster kept calling my name.  So I ended up selling all of the Strat parts.  I was able to break-even.

image_2

Over the past month or so I've been roaming local guitar shops, various web sites, eBay, Craigslist and pawn shops. I've collected enough parts to build a modified '52 Esquire. My goal is to give it a relic type finish. We shall see what happens.

image_1

~Andy

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

Programming Inexpensive Lighting Controllers- Part Two B: Programming Scenes- Strategies

Obey 40 and Obey 70 Scene Buttons A Theoretical Problem

Say for example I just bought two really sweet new moving head lights. I can't wait to break them out. I've been asked to DJ a wedding reception in the garden atrium of the local swanky hotel. It's going to be a great night. 9:00 PM rolls around, a lot of the older guests have left, and it's time to start the dance party. You fire off the new dance mix of a Taylor Swift song, the dance floor is packed and you fire up the movers! They're moving all around, changing colors, and flashing.  It's like a mini club, and everyone is dancing. It's great! You see the bride's Aunt Wendy doing some sort of worm meets electro-boogie dance in the middle of the floor. You suddenly realize she's actually severely allergic to peanuts and is having some problems. You need to get control of your lights quickly.

We'll explore a few strategies for setting some scenes on your controller. These are just a few ideas I've used that work. They're not the only way to do things but they might be a good starting point.

Know your event

I have recently had to help a few customers program their Chauvet Obey 70 controller, and lights.  One gentleman was a member of a party band, and another customer DJ's a lot of wedding receptions..  We tried to program some scenes that would generally work well for most weddings for both customers.

The first step is to plan ahead. Know your event. Know when things are going to happen. Most wedding receptions follow a similar order. Things start off pretty chill, the bridal party is introduced, there's a father/bride dance, food might be served, a cake is cut, flowers are thrown, and at some point (hopefully) a raging dance party ensues.

Considerations

Since most wedding receptions start off pretty low key, keep the lighting low key.  Let your light show build as the evening builds.  This sets up a few things well.

  1. The first dance, father/daughter, etc usually happen early in the evening. If you fire up your movers, lasers, etc during these songs, you risk putting dots of color across the people dancing. Why is this a problem?  It will look weird in the photo album.
  2. If you're playing light dinner music and your light show is screaming "Disco Inferno" you might make someone nauseous. The other thing to consider is that your mirror ball might loose it's impact if it  runs all night long. Save it for a special slow song.  You wouldn't play Abba's Dancing Queen all night would you? Then consider cycling lights on and off throughout the evening.
  3. If you're using a controller that has multiple layers or scene banks, consider putting a "white" scene on each layer, with your movers pointed straight down at the floor or up at the ceiling. That way if Aunt Wendy goes into a fit, you can quickly bring up lights, pull the movers off the crowd and out of people's faces.

~Andy

Hey! What's that Sound? The Smaart Solo Bus

I just picked up Smaart v.7 from work. Let me just start off by saying, "man, what a powerful tool."  The ways you can use it are almost limitless, really depending on your own knowledge. My Solo Buss Idea

I was sitting at lunch today eating a sandwich and thinking about what sort of things I could use Smaart to tell me. Then the idea hit me- "What if I could have a frequency response trace for every single channel on the console?"  Then you would ask, "Why?" I could see problem frequencies on each channel. I would know why Sarah's voice sounds so honky. Or why the kick drum kinda rings out and lacks definition.  I'd have a plot of every channel.

And that might be great but if you have a 32 channel desk then that a lot of information for you to put on the screen at one time. Not to mention you'd be taxing the processor on that poor computer. There's a lot of math behind this information. So I decided there's a really easy way to set this up on a DigiCo console utilizing direct outs. So here's the basic steps. I'm just going to assume you kinda understand setting up Smaart, as well as a Digico console, mainly for time's sake.

  1. Set up a Digico UB MADI or DigiGrid MGB on your computer running Smaart.
  2. Set up a channel or a few channels on the console for measurement mics and route the direct outs for those channels via MADI to the UB Madi or MGB
  3. Set up direct outs on your solo bus to run out via MADI to the UB MADI or MGB as well.
  4. Set up Smaart to use the UB MADI or MGB as it's input device, route the signals, and  label things in Smaart appropriately.

And boom you're done!  You now have an RTA on every channel on your console. Quite handy for ringing out monitors, or finding anomalies very quickly. Who knows, maybe you'll figure out how to keep your mix phase and time coherent all the way through the console What you do with this information and how you process it is up to you!

~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