Features comparisons - an AudioBox & SFX and a taste of the future: SoundMan-Server

UPDATE: What follows below was true at the time it was written, almost 2 years ago - but a great deal has changed - see the last few paragraphs of this post. cbu (11/27/2007)

This could be subtitled: Keeping your sanity......and the future of serving sound on-a-budget!

An AudioBox has tools to program it that run on both the Macintosh as well as Windows. I will concentrate on the Windows tools since that is what I'm presently using.

Both sets of tools are available on the Richmond Sound Design site [ windows Macintosh ] as well as the Harmonic Function, Inc. site. Please note, the Macintosh programming software expects an AB64 while the Windows software (ABEdit) will program and control all versions of an AudioBox (1616HD, AB1616, and AB64)

In talking with Charlie, it's my understanding that the ABEdit software started out as just a programming tool but has since morphed into both a programming tool as well as a means to run a show.

Here's the links to ABEdit manual for the version that is specific to a 1616HD and AB1616 - and this is the link for ABEdit manual that is specific to an AB64.

In this section, I'd like to contrast and compare two products for playing back sound and doing "show control" in a live theater environment. Along the way, I'll indicate things that were show stoppers for me in SFX as well as what was most difficult in doing my first show using an AudioBox.

Sidebar:
Please note that "Show Control" in the venue I'm discussing (Community Theatre) is not on par with Show Control at Disney or a cruise ship or large entertainment venue. But "Show Control" in this venue is a big savings in time, effort, and helping us give each and every audience the same experience. [well, actors not withstanding....]

Prior to SFX and later, the AudioBox; we always did doorbells, phone rings, lightening, electrical discharges (Cuckoo's Nest) and the like manually.

SFX and later the AudioBox, gave us the ability to use MIDI controlled relays ( MIDISolutions R8 ) for effects we had previously used a Stage Manager or Lighting/Sound board operator to execute. FWIW, the R8 is pretty quiet.

For phones and doorbells, I trigger them directly; for lighting and mains operated bells, I go thru a solid state relay (SSR) sized to withstand the devices current and voltage. I use a stock RadioShack 12V power supply ( 1 amps @ 12VDC P/N 273-1776) to provide power to the SSR via the R8. Any 12VDC PSU would work.

The version of SFX I'm experienced with is SFX ProAudio V5.6 build 9 which I upgraded from SFX standard - JUST to get the audio looping bit. [Note: as of November 2007, SFX has new, completely re-written versions out]

Looping comes standard with an AudioBox - ANY AudioBox.....

Also, there are many effects - such as lightening, that require the use of sound effects and lighting effects at the same time. The AudioBox made it very easy.

end sidebar

I promised this would be a comparison of SFX and the Audio Box in terms of what it meant to our venue - it will be by no means an exhaustive comparison - there are others who use one or the other products (or both) daily in their work (for hire). If you're interested in pursuing the archives, then join up with the Theatre-Sound list, The Audio Box mailing list, the ShowControl mailing list, the ShowMan mailing list, and the SFX mailing list. Theatre-Sound, ShowControl & the SFX lists are quite active.

Both SFX and the Audio Box are just tools. Pure and simple.

The first show I designed with SFX was The Taming of the Shrew and had only scene change music and the like - not really a test of SFX but a means to ease me into the idea of computer controlled playback systems.

Not very fancy except I worked with a composer that was quite taken with SFX's ability to seamlessly loop his music for scene changes and the like. In fact, he gave me two versions of each selection - one that could be looped (and faded) and one that could be played to its natural end.

He wrote the music in Finale, saved to MIDI, emailed it to me, and I used Wavemaker III to render his tracks to wav files. I'm no musician, but although he is, neither one of us could afford studio musicians to record his tracks - so Wavemaker was our option. There are other rendering programs out there too - Wavemaker hit the spot at the time.

Shrew wasn't much of a test - but it was just practice for my next project - The Diary of Anne Frank.

On that show, I designed the Lighting and Sound and had a seperate board operator for sound while I was the board operator for lighting. [Usually, I'm both - "left hand = sound, right hand = lighting" - I said it was community theatre, didn't I?]

Anne Frank was much too complex for me to run both even with SFX. Aside from the normal soundscapes that this show requires, we had phone rings that were automatic and to the proper cadance, alarm bells that were very intrusive and some Foley that was pre-recorded but timed on the actors movements.

The phone rings and alarm bells were nothing more than 2 relays in a MIDISolutions R8 programmed to activate in a particular cadence on cue. In SFX, I set each device in its own list and then triggered that list from my main list.

When we did the show about 10 years ago, sound was a nightmare - Fostex 4-track cassette playback system with additional cassette playback units - most of the soundscapes were hard programmed - you couldn't slide them on the timeline at all....

So, what did SFX get me?

On Anne Frank - the ability to fine-tune electro-mechanical noise makers (phone rings, alarm bells), to wash the audience in sound-pulling them into the environment, to have multiple effects running at the same time and at a pre-determined level and location, to have a repeatible sound experience for the audience, and to integrate sound into the overall experience.

Could an AudioBox done the same thing? Yep! and more. But I didn't have one yet.

Forgot to mention, on Anne Frank, we had to integrate a video player at the top and end of the show - For the closing (the tape was the fill for the actors to get into their "curtain call" tableau) I rolled the tape while my sound op gave me a countdown as to how much time (SFX can show the remaining time or elapsed time of an audio file) we had left on the last scene change music - the idea was, at the proper time, the tape was up to speed and stable, the big screen TV was stable (on-a-budget, I said), and the audience was hearing the last of the music.

It worked most of the time - very stressful as I still had lights to contend with.

The last show I used SFX on was "The Mystery of Irma Vep" and that's because I had not allocated myself enough time to learn the AudioBox to the point I thought I could design a show on it.

Here's something to keep in mind about a SFX rig and an Audio Box.

SFX is "easy" to use - doesn't require much study to get going - but once you get past some of the bits that make it easy to use, I found that there were some things lacking. I originally purchased the Standard version - but soon upgraded to the ProAudio version just to get audio looping.

Some people say the fades aren't very smooth - it depends on your sample depth (16 bit vs. 24 bit) - and how low of a level you need - I've never had any trouble.

But what became the break point for me on the SFX vs. Audio Box front was the fact that I could have live inputs as well as hard disk playback on an Audio Box and couldn't on a SFX rig. The Audio Box's EQ, delay, 16x16 matrix were just icing on the cake.

Most shows don't require live inputs in our venue - except 2 types - musicals and a charity show we do to help out the local Salvation Army at Christmas. See http://www.zct.org/0708productions/AT17.HTM for this year's show.

In the past, we've run a feed from the keyboards to a couple of QSC 850 amps feeding on-stage monitor speakers and tried to balance the house and stage sound so the audience was not getting the majority of sound from the stage.

Our "pit" is in the lighting and sound booth. It's not really a pit nor a booth. It's what would be the balconey (choir) in a church, since that's what the venue used to be...and we're doing theatre in it....

SFX didn't have any kind of live sound ability unless you used an outboard MIDI controlled mixer such as a Yamaha 01V and the like. Which I don't own nor does the venue...Heck, the venue only owns lighting....and only some of that too.

The charity show we do each year starts out with open auditions where all comers are cast - some are seasoned actors, some are trying this out for the first time and some are in the middle. It has become very clear in the last few years that some form of musical support (feed) on-stage is needed to keep them together during the show.

This year, I used an Audio Box AB1616 to take a feed from both keyboards, run each to their own input, send the output from each keyboard to both the right & left of the on-stage monitors (a mono feed split to both speakers). We were able to mix both keyboards in such a way as the on-stage feed gave them what they needed to hear (and were used to hearing) and the audience didn't hear the feed from the stage but heard only the pit.

On top of that, I had spot sound effects, video segments with audio, video segments with a dry sound track covered by a pre-recorded track played from the Audio Box, as well as the normal pre-show and intermission music.

Alas, the video player was not controlled by the Audio Box but it could have been. Maybe next year....

Here's an eyeopener on the diffs between a SFX based show and an Audio Box based show - SFX allows you to be sloppy in your show documentation and still succeed.

An Audio Box will soon show up your documentation shortcoming and stop you in your tracks.

SFX can be considered a "streaming audio" playback system. You don't have to worry (within limits) how many cues use what groups to playback your sound. SFX supports 4 groups with each group made up of a stereo pair. You can have all your cues play via Group 1, one on top of another or layered so that a cue starts to playback before another finishes.

SFX forces you to use a stereo group hardpatched to a stereo output amp.

In a way, this is easy for newbies but deadly when you transfer to an Audio Box where there are a finite number of playback channels

In an Audio Box, you have a finite number of live or disk playback channels that can be patched to ANY output device (amp/speaker) or, in the case of an AB64, CobraNet desitination.

Now, all of a sudden, you HAVE to clean up your paperwork so YOU know what playback channels are in use and when. Lucas Cooper in this link indicates a method of working that suggests you allocate certain PB channels for specific tasks. It's not a bad idea.

A SFX rig can be limited in the number of audio files it can playback in synch - because of shortcomings of the user's PC hardware and MicroSoft's DirectX sound engine.

An Audio Box (any flavor) can easily playback any and all audio in perfect synch.

UPDATE - November 2007
It's November 2007 and alot has changed - AB1616s, and HD1616 are no longer in production and are falling from favor by us little guys because it's getting increasingly harder to get a SCSI interface to work under any of Microsoft's windows OS.

In May of 2007, I lost the ability to communicate via a SCSI card in the PC to the AudioBox AB1616 - I did that show with just spot sound.

Enter SoundMan-Server and SoundMan-Designer - which is what my next post will be about -and it's SO exciting - Theatre's that have practically NO budget can now have 4 channels of playback (equal to 2 stereo pairs), two live inputs, a stereo pair of outputs AND a full featured ShowControl system for FREE!!!!! - no license fee at all!

To get a taste of it, download the demo from here: http://www.richmondsounddesign.com/softw.html#SMD

Full requirements are on the website http://www.richmondsounddesign.com/sm-dref.html

And yes, you'll find a link to this blog there too.

Next installment (and it won't be 2 years either) will take us from ShowControl of Alcorn-McBride DVM2 MPEG-2 video servers plus sound effects for The Laramie Project to the despair of no AB1616 working with the SCSI port on our sound PC (May 2007) to running a full blown SoundMan-Designer system for a production of Foxfire (October 2007) - with many tracks of layered sound, backing tracks for live musicians and a whole lot of effects that added to the audience's experiences.

Back Soon!

- Carl

Learning Curves - Steep they are - But worth it!

An AudioBox (any flavor - 1616HD, AB1616 or the latest and greatest - AB64) is a matrix mixer that is completely under software control and responds to Midi Show Control commands (MSC), the full spec is located on Richmond's site as in here. This is for V1.0 of the spec.

A better description of what an AudioBox is from www.kiss-box.com : It's a "Digital Audio Playback, Signal Processing, Sound Placement and Spatialization fully utilized with the AudioBox virtual matrix and hard disk playback/recording system". Please note, the AB64 allows for recording as well as all of the preceding.

The venue I usually work in does not have nor need any live sound reinforcement system (we seat 92 and the stage is considered a "black box") and so our designers have limited experience in using a common mixing system.

That said, the most difficult part in programming the AB1616 or ANY AudioBox is to understand that some programming has to be done to get ANY output at all!

Let me back up a moment.

Any RSD AudioBox supports a combination of live inputs and disk playback channels linked to a similar quanity of output channels. The AB1616 I have supports 16 live or disk based (or some combination that can be changed on a cue to cue basis) "playback" channels (best to think of them as sources of audio that may or may not be a stereo pair, and, that can change on a cue to cue basis too) that can be routed (sent) to 16 output channels.

On top of that, you can adjust EQ, delay, fade up/down/pan levels all on the fly - depending on the needs of the show, under program control (using ABEdit and MSC) - or by using a CM Labs MotorMix, in conjunction with the RDS AB, you can make similar adjustments LIVE!

Sidebar: Ever adjusted all your playback levels perfectly in tech, only to find with an audience in the space, your levels are too low? With SFX, there is no way to adjust ALL or ANY of your output levels on-the-fly. With a RSD AB and a CM Labs MotorMix, you can assign submasters to various inputs & outputs and adjust them independently or in groups. For those of you familar with lighting boards, the RSD AB subs work just like the subs on a lighting board - they raise or lower all assigned channels in proportion.

Back to what is an AB and it's learning curve.

Understand this [and it's mind boggling, and stressful if you're doing it on a deadline] - these are what you can program (and in some cases HAVE to program) to get your expected output:

• Submaster assignments (there are 32 subs that can be assigned under program control)
• Submaster zero point [the AB is a unity gain box, so it doesn't boost the signal but can cut it, you can setup your subs so that they let more of a hot signal in or provide cut too. It's all in how you adjust the gain of the box]
  
• Input assignment: Live feed or disk playback
• Input level, EQ, delay
• Crosspoint: level, input & output
• Output level, EQ, and delay
• Fade up or down speed and stop point on all Inputs and Outputs.
• BTW, the EQ and delay can be adjusted in real-time.

ABEdit is the free programming and playback software provided for the Windows platform.

Free Windows and Macintosh programming software is available on Richmond Sound Design's site. I'll talk about the ABEdit learning curve in a later part. BTW, this all has a happy ending - at least for the ZCT's recent show: The Haunting of Hill House.

Note: It's also good to have all your effects gathered up, approved by the powers that be BEFORE you begin programming the box.

Because if you don't, it's pure hell. Making a stressful time even more so.

How many AudioBoxes are there?

Previously, I've said there are 3 kinds of AudioBoxes (all are available from RSD but only the AB64 is available as new):

• First Generation: a 1616HD which sports 8 live inputs, 8 disk playback inputs and 16 output channels, both MIDI and SCSI inputs and uses SCSI from a PC or Macintosh to load wav files to the internal SCSI disk storage. MSC is the control software command language.
  
• Second Generation: an AB1616 which sports 16 live inputs, 16 disk playback inputs (which only 16 can be active at any one time) and 16 output channels, both MIDI and SCSI inputs and uses SCSI from a PC or Macintosh to load wav files to the internal SCSI disk storage. MSC is the control software command language.
• Third Generation: an AB64 which is the most flexible AudioBox to date. Instead of using a SCSI interface to load wav files, it is a box that is network aware and uses TCP/IP to transfer audio files and provide control. It also has dual MIDI inputs, outputs, and thru connectors. MSC is the control software command language. It also uses IDE drives and can support 2 drives in a mirror 1 configuration. Nifty, eh?

There are also hardened AudioBoxes available that are more suited to the theme park and entertainment industry. They are custom built for the client and contain, among other changes, solid state disks in place of SCSI or IDE drives and the customer's connector choice for I/O and power.

Major technical specs for all the "stock" AudioBoxes can be had on Richmond Sound Design site as well as Harmonic Function's site.

Back to points that can make your life difficult if you forget about them or OK and easier if you remember:

ALL AudioBoxes use audio files that are at the 48K sample rate. It's the AES standard - see RSD FAQ. No matter how they started out, 48K is how they wind up stored in the AB.

Furthermore, in processing by the ABEdit software, the file is changed to 48K sample rate and split into 2 mono files, a right and a left.

If a file is already at the sample rate of 48K, the ABEdit software only splits it into a right and left file, assuming it started out life as a stereo file.

I use Sound Forge to edit my audio files, and to rip CD tracks and to process these tracks further. Sound Forge also supports the batch conversion of files to another sample rate such as 48K - you can do that to save a step in the ABEdit software. I didn't on this trip because I wasn't very organized - or at least not organized like an AudioBox show needs to be.

Files are transferred from the computer to the AudioBox using SCSI (if the AudioBox in question is a 1616HD or AB1616) and via a network connection (TCP/IP) if an AB64.

Note: It's a good idea to develop a protocol/procedure that normalizes your files BEFORE you transfer them and resample them in the AB. In Sound Forge, I usually normalize music files to -16db - there's a preset for that and there is a preset for speech (voiceovers) too. Sometimes, I drop the level even more.

Here be the reason - yes, the ABs have analog inputs and outputs but they process audio in the digital domain. Too hot of an input signal WILL result in distortion - it's a crackle that coincides with the peaks of the audio signal. Reduction of the channel input level (in the AB) may alter it but won't get rid of it.....

You want a procedure that nomalizes the signal before it's transferred to the AB. You also want to consider whether or not you're going to adjust EQ in the AudioBox - some EQ will increase the level of the signal resulting in digital distortion - it may be a trial and error process. Most likely will be.....

You can adjust the level of the file that gets transferred by the ABEdit software at the time you resample and split to right and left channels - BUT - this is global for all files transferred in that session. If you normalize the file in your editor, that's your best choice. Then take the default level in ABEdit when you transfer the files.

Here's the gotcha (one of many):

The AudioBox is a unity gain device - you can cut levels, but not boost them. Depending on how you set up your overall gain structure, you may be able to increase the I/O levels (because you've been cutting the level all along) via the submasters. But you have to take into account what is the loudest signal you need in your venue and make sure your gain structure allows it. Trial and error again...
------------------------------------------------------------------------------------------------

Back to basic setup and what works and what doesn't quite....

Your computer needs to be a PC with a SCSI card or interface if you're running the ABEdit software.

SCSI cards and OS that I have tried are - W2K pro with a SIIG AP-20. I suspect a Kouwell KW-910U will work too as they look identical right down to the contents of the manual. These are PCI cards that you have to open up the computer to install. It's very easy and safe.

I had success with a Belkin F5U115-UNV USB to SCSI adapter under WinXP home; however when I moved it to the W2K machine, it crashed the box. Twice. 1st time I knew what had happened, second time I didn't. Time to install a "real" SCSI interface card. No crashes to date with the SIIG AP-20 on the W2K machine.

Audio (wav) file transfers from the PC to the AudioBox are much faster using a SCSI "real" interface card compared to the USB card.

I'm in the process of evaluating a RATOC firewire to SCSI interface for use by the AB1616 and I'll let you know how that turns out. So far, on a Dell running WinXP Home, it doesn't look promising......I haven't tried it on my sound machine running W2K yet.

Some Working procedures:

[at least, this is what I used during my initial voyage]

Basic routing on the AudioBox I used went like this:
So my content approvers could hear the effects via the amps and speakers they would hear the final effects -

Sound computer has an Echo Audio Gina24 2 in 8 out sound card in it which I routed Audio Out 1[Left] & 2[Right] (set up as the default output in Sound Forge) to the AudioBox inputs 1[Left] & 2[Right]

I then patched AB Input 1 to AB Outputs 1, 3, 5, 7, 9, 11, 13, 15
and then patched AB Input 2 to AB Outputs 2, 4, 6, 8, 10, 12, 14, 16

I created a cue that I could recall at will to recreate this routing w/o interferring with the real cues of the show.

Now, I also created a cue that said patch AB Inputs 1, 3, 5, 7, 9, 11, 13, 15 to AB Output 15 and another one that patched AB Inputs 2, 4, 6, 8, 10, 12, 14, 16 to AB Output 16.

This cue I left in my setup or turn-on cues - this let me plug in a headphone amp into outputs 15, 16 so that I could monitor the AudioBox on headphones.....letting me program the box w/o having to program it and run it "out loud" in the venue (or at home). I could also listen to Sound Forge this way too, helping the editing process.

ZCT only has 8 channels of amps so dedicating AB Outputs 15, 16 to a headphone amp is no big deal....

But I'm getting ahead of myself. This programming stuff.

Prep and approval of audio files BEFORE you program the AudioBox!!!!!!!!

You HAVE to have the idea that all sound content is appoved by the powers that be BEFORE you start to program the AudioBox.

Edit all your content, for level, looping, time, whatever and save it to disk with a name that makes sense - perhaps it names the cue page or what the cue is or SOMETHING that tells you when you're looking to place this file in the AudioBox what it is.

Also, make sure you still have a copy of the original file, and the newly edited version as well as all the ones that the director didn't like (because, at some point, he or she may ask you to put in the one they didn't like 3 days ago but do now.....)

When you are satisfied with the sound content for a particular cue and you can get the powers to OK the content, THEN transfer the file to the AudioBox.

If you don't do this, then you'll be caught in a loop where you have to constantly re-edit and re-transfer the sound content over and over again. Under the stress of Tech or the Direcor or TD hovering over your shoulder will make you wish you still used razor blades to edit audio.

Maybe you're more organized than I am - but let me tell you, it isn't fun to have to re-edit a file and re-transfer it because the loop isn't perfect (or, more likely, they didn't need it looped but do now....because something with the set design no longer works and they need fill)....

OK, you've got your content, you have it approved, they love it and you have it transferred to the AudioBox.

Now what? You're not done by any means but tomorrow is another day.

Next time out - some more basic programming and working procedures. I have to collect my resources on this so that it's more concise than what I've written so far.

I'm hoping to go into what the ABEdit manual is not as well as get permission to use some of Richard Ingraham's content from his web site (or link to it anyway). I already have permission to link to RSD's site - they have some very good papers on-site that I think actors, sound designers, directors, and other tech personnel can benefit.

Thanks

Introduction - Making Waves with Light and Sound

I retired from my day job on Halloween, 2005.

I had worked for the State of Ohio department of MR/DD for 27+ years and it was time for a change. I've been an AV/Media Specialist, a Systems Analyst, and lastly, a Network Administrator.

But what I really love to do, is Theatre. In fact, one of the major reasons to retire (plus, I got a buy-out ) was that the job was getting in the way of family and theatre.

So, it was time. I guess.

But what has started this blog is something new that I've tackled to learn - As my brief bio from Working (the musical based on the book by Studs Terkel) talks about, I've been an audio editor since the days of splicing tape, an EdiTall editing block and reel to reel tape.

In the 70's, I also learned to edit video tape with control track editors....crude to say the least.

I had been using reel-to-reel tape for quite some time (since 1968) to run spot sound for shows - spot sound is 1 sound at a time - most sound effects such as doorbells and phone rings were done live.

But in 1997, I ran reel to reel for the last time in a production of One Flew Over the Cuckoo's Nest. The machine was dying and I had to find a better way. That better way, was a ShortCut editor by 360 systems and made it possible to do much more than was possible with tape. I used the ShortCut for a few years and then moved up to a software product called SFX ProAudio from Stage Research in Cleveland, Ohio. Now we could have up to 8 stereo pairs running at the same time as well as MIDI controlled relays for bells, flash pots and all manner of devices.

But something was lacking. In the late 90's, I had heard about a device that was a software controlled matrix mixer called an AudioBox produced by Richmond Sound Design of Canada. Charlie Richmond, the founder, was/and still is very active on a number of sound and theatre lists that I subscribe to - and it was Charlie who, in the late 90's suggested that I demo an AudioBox.

Well, let's get real.

If I had demo'd the box and liked it (which I knew I would), I'd have felt obligated to buy it. And I just couldn't afford it. I'd have had to personally purchased it as the community theatre I volunteer with just wouldn't have been able to afford it at all!

So, I've made do with SFX (not a bad product, really) and the ShortCut (good product here too) and just lusted after an AB1616....

Fast Forward to June 2005 - The AB64 is now out and has been for about a year. I see where a used AB1616 is going for $2,000 USD and I think - nice retirement present!

So I ask Charlie (isn't e-mail great!) if an AB1616 would be available and, if so, could I purchase on a lease/purchase plan. Long story short, yes, there is one available (Purdue University just upgraded to an AB64 and so I purchased their AB1616 in July) and yes, he would do a lease/purchase arrangement.

Cool!

Next part will detail some of the learning curve issues as well as some of the benefits of running sound with ANY AudioBox, but specifically an AB1616.