Good work, Dave!

The impetus for this post came from a simple question; is it better to hit the pre-amps hard then turn down at the main output, or run the mains up around unity and dial back input gain to get the SPL you want out of the system. As a general rule (there is an exception, which I’ll detail in a minute), I would argue the former is the correct (or at least better) method, and here’s why. Most pre-amps sound best when you hit them pretty hard (at least up to the point of clipping, which is too hard). By running your pre-amps hard—and by hard I mean around -6 dB full-scale on a digital board, or within 6 dB of clipping on an analog board—you are maximizing your signal to noise ratio. And for some reason, they just sound better. Keep in mind, that’s a general rule, your mileage may vary. Now, it’s quite possible that if you dial your input gains up so that all your pre-amps are running high, your overall system level will be too high. That’s when you would lower your main level to compensate. This method will keep the signal to noise ratio high throughout the mixing chain, and will attenuate the signal at the last possible moment.

Before we get to setting up the gain structure, let me lay out my goals in for the process. First, I want to maximize S/N ratio, and use up as many of the bits in the analog to digital (A/D) conversion process that I can. Keeping the input level high meets both goals. Second, I like to mix with my faders around unity. Mixing with faders at unity is another key ingredient to good mixing. The fader resolution is highest right around unity, so you can easily make small adjustments. If you try to mix with your faders at -20, a slight change in fader position might yield a 3-5 dB change rather than the 1-2 you actually desire. Finally, I want to be sending a very solid signal out of my mixer to the processors for the same reasons (only in reverse) as the first point. That’s why proper system gain structure is important.

Here’s how I would approach the process.

Gain setting in the digital world
For each input channel, I would have the musician play their at their loudest level. I then dial up the input gain until I’m within about 8-12 dB of full scale (minus 8-12 dB on the meters). I like to leave a little room for the musician to play louder when the lights go up (they always do). Many digital boards also have a trim (or attenuation) control in addition to the input gain. I use my trim to dial the level back to where it should be in the mix with my faders at unity. Because I’ve gained my entire system properly, my main fader is sitting at unity as well, and all is right with the world. As I am using VCAs to manage groups of faders (drums, guitars, keys, bgvs, etc.) and those live at unity as well, at least to start. All of this ensures that my signal to noise ratio is optimized at the A/D stage (just after the mic pre), and my starting point for my mix is faders at unity.

Now, if you don’t have a digital trim control on your board, you have a decision to make. You won’t likely be able to run the mic pre’s hard without having too much signal at some point, so you’ll need to dial the level back somewhere. Of course, you can always turn the fader down, but then you lose fader resolution. A better alternative would be to use a VCA to keep your fader at unity, though that can get tricky. Take a drum kit for example: If you optimize the gain on the kick, snare and hat, chances are, the hat will be way too loud in the mix. But more than likely, you’re using a single VCA for the entire drum kit. So now what? Well, you could break the drums up into zones and use one VCA for each; kick & snare, toms, hat & overheads might work. That way you can pull back the faders at the VCA level (a VCA is really an electronic remote control of the faders), and maintain fader resolution. You could do a similar trick with groups if you have them.

If you’re running short of VCAs, I would break my rule and set the input gain up so that the fader remains around unity for a proper mix. Audio is a lot about compromise, and in this case I’ll give up absolute input S/N to run my faders at unity. I have found that to be the wiser trade.

Gain setting in an analog world
Really, the process is much the same, though you are much less likely to have a trim control after the gain control. In that case, the same rules apply as a digital board without a trim knob. You still want to have good input level coming into the channel (for the most part), then turn it down as needed later in the mixing stage. You also want to keep your faders running around unity. Make the trades where you have to.

In either the digital or analog world, what you don’t want to do is under-drive your mic pre’s and have to add a lot of gain down the road. Sure you can push a fader up for a guitar solo, but you don’t want to regularly run your input faders at +8, your groups at +10 and your main at +5 because your input gain is set too low.

The exception to the rule
Now, all of this assumes you’re running on a professional grade mixer that has a mix structure designed with proper headroom. If you find yourself mixing on a Mackie or Behringer (or similar music store brand), chances are you’ll run out of headroom in your mix bus very quickly. If you set input gains on a Mackie the way you should, when all those hot signals hit the mix bux, it’s not going to be pretty. The busses quickly saturate and you lose all sense of dynamics. In that case, you need to really keep an eye on your overall output level and run input gains down accordingly. This isn’t a dig on cheap mixers—you can only expect so much for what you pay for them—it’s just reality.

That’s a quick guide to setting up your gain for an input channel. As I mentioned earlier, if you go through this whole process only to find that your overall SPL in the house system is either way too loud or way too soft, you have some work to do at the system processor or amp level. But that’s another post…

One thing that has excited me more than almost anything else is the rise of a technical artist’s community over the last few years. Technology itself has made it possible for church techs to connect with each other, even if we’re scattered all over the country. Now, I’m clearly a Twitter fanatic (check my stats…), but that platform alone has made it possible to connect with a great group of people (not just church techs) all over the country. Often, I find myself DM’ing or @replying to people even locally rather than texting; sometimes it’s faster. There are so many people that I’ve gotten to know over Twitter, and a good number of those online relationships have turned into great friendships. Sometimes we even get to meet in person (or IRL, In Real Life, as they say).

While online relationships are great, and I value those that I have, I think we also need to be in community with other people around us. It fascinates and puzzles me when I hear about a local church TD who doesn’t really have any desire to get to know other TDs or spend any time with them. I really don’t know why that is. I know we’re all busy–I have a to-do list that grows by the hour–but still I make the time to meet with other TDs around here. The reason is simple; no one else gets me like another TD. We all know ministry is hard, and technical ministry is really hard. Typically, there is just one or two tech people on any given staff. That means there is no one else in your church who you can complain to, bounce ideas off of or talk things through. Few people have any idea what we even do, let alone understand it. That’s why at least once a month, I try to have lunch with a TD from my local area. Sometimes “local” means a 45-60 minute drive. But you know what? It’s totally worth it. I know I walk away from those meetings encouraged, and I’m told those who have lunch with me are encouraged also. And it’s not because I’m this great encourager; there’s just something about spending time with another TD that makes our own troubles seem smaller.

Another high point for me is the local SoCal CDTRT meetings we’ve been having. This past Monday marked the third “official” SoCal CTDRT Meet Up, and we once again had over 15 guys, some of whom drove 2-3 hours to get there. Why did they do that? Because they recognize the importance of meeting with other TDs. Sure, we played with some cool gear and got free lunch. But I’m pretty sure that’s not the draw. The reason they all came is the conversations that happen around the tables at lunch, and all afternoon. I’m an introvert, so those meetings are draining for me, but I love them. I have gotten to know some great guys at these events, and I want to spend more time with them. So we’ll do another in January.

So what can/should you do if you’re a TD? First, seek out other TDs in your community. Look in the CTDRT directory and see who’s around you. If you’re not a member of the CTDRT, join. If you can’t find anyone, start calling around to larger churches and see if you can figure out who the tech guy is (websites are great for this…). Go out to lunch or coffee with them. Or stop by and visit to check our their facility. Maybe even stop in for a service if you can swing the schedule. Perhaps you need to be the one to organize a local CTDRT meet up. It doesn’t have to be fancy. You don’t have to bring in gear to demo or even find a sponsor to buy lunch. Just meet at a restaurant and have lunch (that’s how the SoCal meetings started…). The bottom line is you have have to make the effort. Pretend you’re a jock and “Just Do It.” Trust me, you won’t regret it.

As always, you can listen to the audio here using the player below, download the file from the link or subscribe to the iTunes podcast feed. Thanks for listening!

Right click this link to download the file.

To recap, we have stopped using alkaline batteries almost completely. I say almost because I still have a few around for the odd guitar player with a dead 9v (ProTip: If your guitar takes a battery, have an extra one or three in your case, OK?), or for the occasional metronome or other device that shows up on stage unexpectedly. However, for our regular production equipment, we’re completely rechargeable.

We went with a variety of batteries in our system. For the lead pastor’s UR1M, we have two sets of Powerex AAA batteries. Those live in a dedicated 4 cell charger that charges them nice and slowly. There has yet to be a time when the “fuel gauge” on that mic drops below 4 of 5 bars on a weekend. I have a set of eight Ansmann, a set of eight Sanyo and two sets of eight Powerex cells for our UR1s and UR2s, as well as the PSM900 receiver. The AAs get more run time on a typical weekend than do the AAAs, but we’ve never had one drop down below 3 bars. I also bought a bunch of Powerex Imedions for use in our MightyBrite music stand lights. Those are LED-based lights, so the power draw is quite low. The Imedion is a 800 mah, low self-discharge design. I’ve found we can easily get three weekends of stand light use out of them before needing a recharge. We could probably go four, but if we use the same stand lights every weekend, they tend to start getting a bit dim on weekend four. Since it really doesn’t cost anything to recharge them, we do.

We’ve established a really straight-forward process for managing the batteries. I have the capacity to charge 24 AAs at once, so we leave 24 batteries on trickle charge all week. Come Saturday afternoon when we’re setting the stage, we’ll pull batteries out of the charger and load up the mics. Depending on the weekend, we’ll use between 10-18 cells. Saturday night, the monitor engineer collects the mics (and the PSM900) and puts the batteries back in the charger. We always charge using the soft charge mode, which charges at a rate of 500 mah. Sunday morning, we repeat the process. With Alkaline batteries, we would typically try to stretch the batteries for a full weekend (though, they often didn’t make it). With our current process, we may have added a few minutes worth of battery change time to our weekend routine.

For the Imedions, I have a recurring task set up in my Associate TD’s to do list every three weeks. All told, it takes about 10-15 minutes to cycle the batteries out of the lights, into the chargers and back. Since that’s only once every 3 weeks, it’s not a big cost. I have more Imedions than we actually use, so if someone leaves a light on overnight, we’re not in trouble.

A Few Observations
As expected, the batteries are working very well. In the past six months, we’ve not had a single mic go out due to batteries. In fact, none have even been close. We have Wireless Workbench set up at FOH to monitor the health of the packs, and it’s gotten to the point where I don’t even think about them anymore.

One thing we have noticed is that the AAs are a little bigger around than Alkaline cells, and that diameter varies between brands. Ansmann are the smallest, followed by Sanyo, then Powerex. The Powerex are a really tight fit in the UR1 bodypacks, and I’ve taken to using Ansmanns in those. In the UR2 handhelds, all of the cells fit, but removing the Powerex cells takes a bit of creativity. I’ve found if you pull the exposed battery out quickly, the force of the spring is enough to eject the other one enough to remove it. Go slow and you’ve got your work cut out for you.

After six months of heavy use, that’s about the only drawback we’ve found to them. Given that we’ve kept over 600 AAs and AAAs out of the landfill and saved over $250 in that time, I feel pretty good about the switch.

As I said, next month I’ll be re-testing the batteries I tested in March and we’ll see exactly how well they’re holding up. From a practical standpoint however, it’s been a great success.

The iPad gets me a lot closer to the action.

The Connections
Our FOH location is far, far away in a balcony. It’s a solid 45 second walk from the main seating area to the FOH location (and involves going out into the lobby, up 2 flights of stairs, back through the second floor lobby, through the balcony and into the tech booth; brilliant I know…). As that location is so far removed from what the people on the floor actually hear, remote mixing has been somewhat of a longstanding desire.

As of this writing, it’s not possible to directly remote into a digital mixer (though I’m hoping that DIGiCo will one day consider adding RDP to the OS…). Thus, an intermediary is required. In our case, we have a MacMini running XP (Bootcamped) running the SD-8 remote editor. The SD-8 editor is nice in the sense that it can be configured to control the audio engine as a separate remote, without effecting the surface. In other words, someone could be mixing on the surface, selecting channels or pfl, adjusting EQ or what have you, while the remote could be doing the same thing. They only bump into each other if both try to adjust the same parameter at the same time. You can also set the remote computer up as a complete mirror if you like.

Anyway, once I launch the SD-8 remote on the Mini, I used an app called Desktop Connect on the iPad to Remote Desktop Connect into it. I like Desktop Connect because it features auto-discovery and will do both VNC and RDP. That means I can very quickly switch back and forth between the Macs and PCs at FOH and control the whole system from one app. Desktop Connect also uses gmail to allow you to sign in to any properly configured machine from anywhere. I haven’t tried that yet, but will at some point.

I’ve used other RDP and VNC apps on the iPhone and none connect as fast or reliably as Desktop Connect. Even when I put the iPad to sleep, it wakes up connected without issue. Score one for that setup.

The Mixing
The band consisted of 3 vocals, 2 guitars, bass, keys and a full drum kit. I had 3 mono wedge mixes, 1 stereo wedge mix and a stereo ears mix. Because of the number of sources and mixes, I started the day off on the SD-8 surface. I got the gains dialed in, set up the monitors and had a rough mix in place. I had also pre-built most of my snapshots for the night.

I then headed downstairs with the iPad. Desktop Connect enables two modes of mouse pointing. One is an offset mode, where moving your finger on the screen moves the pointer. The offset is variable and though it takes a minute to get used to it, it’s very fast. The other is touch mode. I was hoping touch mode would be more useful, as the SD-8 is a touch-based system. However, at least right now, using touch mode was difficult. I couldn’t reliably move faders with it, so mixing that way was out of the question. Touch mode is super-handy for switching screens however, and swapping modes is a simple icon touch.

Impressions
Overall, it worked pretty well. The only real downside is a function of one of the things I love about the SD-8; it’s complete channel strip display for all 12 channels in a bank. Switching from one bank to another means every single pixel has to change, and that took a good 2-3 seconds to re-draw on the iPad. As I was finishing up this post, I started playing with a few things. I have the SD-8 editor running in Win7 Pro on my MacBook Pro under Parallels. When I connected the iPad to my MBP, bank changes were almost instant. What I don’t know is if that extra speed is a function of Win7, the fact that I’m using VNC and not RDP, or the extra processing power of my MBP (the FOH Mac Mini is a CoreSolo 1.5 GHz…). More investigation is clearly in order, however, it does seem that we will be able to speed it up to a point where bank changes are not a real issue.

The use of snapshots was simple, and I mapped the next snapshot and previous snapshots to F-Keys. Desktop Connect has a Function Key menu readily available; press the Fn icon at the top of the tray and the F-Keys immediately appear. I used those to advance through the show.

Touching the glowing Fn button brings up the Function Keys

The thing that did bite me one time was having a snapshot set up incorrectly, which raised the lead vocal mic level to the point of feedback. It took a few seconds to get to the right screen to correct it. Obviously, on the surface, that would have been a half-second. Lesson learned—make sure your snapshots are clean before the service.

Another lesson I learned is to take advantage of the SD-8′s amazing configurability. I should have set up a fader bank with my most-needed channels in it; vocals, guitars, and a few VCAs. That way I could have easily mixed the entire service from one screen, and eliminated a bunch of switching delays. I’ll do that next week.

The takeaway is that mixing from the iPad is almost there. I think with a few tweaks to my workflow, this will be incredibly useful. Is it the same as having 37 faders, 50+ encoders and a 15″ touch screen in front of me? No. Is it a very helpful way to get a sense of what the audience is hearing, tweak parameters and get the mix dialed in? Yes. Will I continue to use it to make our sound 1 better? You bet. My real hope is that DIGiCo will develop an iPad app that will connect directly to the SD-8 without a computer in between. That would be the killer app…

As you know, these days, line arrays are all the rage; every major manufacturer has jumped on the bandwagon, and every integrator I talk to wants to sell me an array. But are they really the best choice for every situation?

On the other hand, there are some old-school guys out there who insist any speaker they mix on has to be a point source box; no new-fangled array for them. But is that always the right option, either?

Those are some of the questions we’ll wrestle with. We may not be able to answer every single situation, but we’ll work through the thought process you might want to take when deciding between the two technologies. It should be a lot of fun.

Tune in on our LiveStream channel at 7 PM PDT, 10 PM PDT, next Tuesday, August 24 and watch the fun live. As always, the audio will be up on iTunes shortly thereafter.

Honestly, I think grey was a good color choice.

Greater Flexibility
Avioms have a lot of things going for them, especially if your band is fairly small. However, once your input count rises above about 20 or so, it’s harder and harder to figure out how to cram all those channels down to the 16 channels offered by Aviom, especially when everyone gets the same 16 channels. It’s even harder if you have a few stereo sources. Sure, you can submix and create stems, but it eats up a lot of mix busses on your console, adds extra work for the engineer and still means everyone has to be happy with the same set of stems.

With the M-48 system, each personal mixer is essentially a 40 channel mixer, and each mixer can be configured individually. Thus, if the drummer wants more discreet control of the drum mics he can have it, while everyone else gets a stereo group mix of the drums. You might give vocalists kick, snare and hat, while the keys player might get the whole kit in his stereo group. The simple fact that you can give each musician 40 channels to chose from, laid out in any configuration of 16 stereo groups they want is enough to tip the scales in favor of the M-48. But it gets better.

Stereo Groups
At first glance, it appears the M-48 is a 16 channel mixer, just like the Aviom. However, just as the SD-8’s channels can be mono or stereo, each of the 16 channels–they’re really groups–are stereo. That means if you want to send a stereo source to a stereo group, it A) stays stereo and B) only takes up 1 channel. And if you send multiple channels into a group, say 8 mics of a drum kit, you can pan them in the software to create a real stereo group of the drum kit. All the musician has to do is turn it up or down; however, the mix is in stereo. This is all done in software on the set up end, and can easily be recalled.

Assigning input channels to groups is as simple as ticking off Xs.

Better Software
Perhaps that should be titled, Software, as the Aviom doesn’t have any. The RSS software enables you to go into each mixer, set up the groups, adjust the level and pan of the channels in the groups and save and recall presets. It’s all easy to use and actually looks like the mixer.

Personal mixing for your personal mixer.

Better Sound
Avioms have never really been known for their great sound. Convenience, usability, and good at reducing stage volume? Yes. But sound? Not so much. In fact, when I was considering a move to Avioms (because the S-MADI Bridge hadn’t been announced yet), every musician I talked with about it was not thrilled because they’ve all had bad Aviom experiences. Now, I’ll admit that a lot of people (churches especially) don’t set them up correctly, over drive them and they sound terrible. I would avoid that problem. Still Avioms don’t sound great under the best of circumstances. The M-48s however, sound fantastic. Rich, warm and full, they’ll be a perfect compliment to the SD-8.

Larger Feature Set
Aviom gives you the ability to adjust level and pan for each channel, mute and solo, adjust bass and treble overall and save presets. That’s about it. The M-48 does all that plus adds a 3-band mid-swept EQ for each channel (plus overall bass & treble), adjustable reverb for each channel, an ambient mic (with volume control), an external 1/8” input jack with volume control, 1/4” and 1/8” headphone outs plus stereo balanced line outs, the ability to record your mix via a 1/8” jack and adjustable threshold limiting.

So that’s my rationale. Once I get the system set up next week, I’ll be able to do some video blogs showing you how this all works in real life. Until then, if you’re considering Aviom, take a good, hard look at the M-48.

The SD-9 is the smallest (currently) in the newer SD line from DIGiCo. The naming structure may be a bit backward; the SD-7 is the big daddy, while the SD-8 fills the middle ground. On the other hand, they’re in chronological order, so I guess it makes sense. The SD-9 is capable of handling 40 input channels. That doesn’t seem like a lot until you read closer and discover that, just like it’s bigger brothers, each of those channels can be mono or stereo. So while it’s not the same as having 80 discreet channels, you can put 80 inputs into it. More than likely, what you’ll find is that having stereo channels anywhere on the board is a huge convenience. For inputs like piano, keys, video, iTunes, or CD, it’s nice to not tie up two channels-and more importantly on smaller-footprint boards-two faders. For the small to mid-sized church, this input count is probably sufficient.

On the output side of things, the SD-9 follows in the footsteps of the SD-7 and SD-8, albeit on a smaller scale. There are a total of 17 groups or aux buses available, each of which can be mono or stereo; how you allocate them is up to you. For example, you could have 16 stereo aux mixes and 1 stereo group (you have to have at least 1 stereo group for main outputs); or 6 mono aux mixes, 6 stereo aux mixes, 2 mono groups and 3 stereo groups; or…well you get the idea. While it seems confusing at first, to me, it’s one of the best features of the console. You can configure it how you need it—which might be different from how I need it. With the SD-9, you’re not locked into a fixed architecture. Reconfiguring the console takes just a few moments, and you’re ready to mix. You also have an 8×8 output matrix, the inputs of which can be anything; a group, an aux or a channel. Again, flexibility is the watchword here.

The standard SD-9 configuration comes with a 32×8 rack; a new model called the D-Rack. Rather than connecting via MADI, it uses a single CAT5. There is an extra output slot on the rack, which can be filled with analog or AES outputs. It’s possible to combine two D-Racks using only CAT5, and using the MADI input on the console a MADI- or DIGI-Rack can also be connected making the pool of inputs an impressive 132. If you were mixing multiple bands at a festival, you could easily swap between inputs using snapshots, reloading shows, or even using the alternate inputs available on each channel strip. It’s also possible to connect using fiber.

The surface of the console is quite small—not even 3 feet wide—and much of that is taken up with the 15” wide touchscreen LCD monitor built right in. There are 24 faders, each with it’s own multi-color 3-line LCD scribble strip and 8-segment input meter. The faders are arranged in two banks of 12, and each bank has four layers. Like the SD-7 and SD-8, each fader is completely assignable. It’s very easy to build banks of input channels, groups, auxes and matrix faders on one bank. Like the bus structure, you lay it out the way you want it.

Whereas the SD-8 has 3 rows of assignable encoders above each fader, the SD-9 has a single row under the monitor. The monitor displays channel strips for one fader bank at a time, something I’ve grown to love on the SD-8 (seeing input gain, EQ, comp, gating and aux sends for 12 channels at once is incredibly useful). The encoders are assignable as a group to gain, lpf, hpf, comp, gate, aux and pan. When in aux mode, a set of scroll buttons select which aux is being controlled. While not as fast as the SD-8, it is quite serviceable.

One thing I didn’t like about the SD-9 is that the monitor is centered over the fader banks, so there is never a time when the monitor channel strips line up with a bank of faders. Coming from the SD-8, I found this somewhat confusing at first, however, after mixing on it for a weekend, my friend Kevin Sanchez reports it’s not bad once you’re used to it. Also provided is a VGA output on the surface for an overview display. I use that as a quick view of all my inputs and outputs on my SD-8, and would do the same on the 9.

Something DIGiCo has done better than almost any console manufacturer is make the software interface consistent throughout the SD line. In fact, were it not for the splash screen and bus/fader limitations, you’d be hard pressed to tell the three versions of software apart. All three boards share the same SFPGA Stealth Processing, there is just more or less of it available depending on the model. Because of bus structure limitations, it’s not currently possible to directly load an SD-7 or SD-8 show onto an SD-9. However, I’m told they’re working on a “Load Partial” feature that will let you truncate the show to fit within the limitations of the console. Still, the good news with the shared processing and software is that the SD-9 sounds just as good as it’s bigger siblings. You also have access to the same effects (in a rack of 4), the same fully parametric EQ (with up to 4 dynamic EQs), built-in compression (with up to 4 multi-band comps available anywhere) and dual insert points on any channel. It should be noted that every input, group, aux, matrix and even the talkback channel has all these features available.

Of course, you also get niceties such as the excellent snapshot system, the macro commands, touch-sensitive faders, dual solo busses and an interface that just screams, “Remote control me with an iPad!” In other words, it’s a lot like the SD-8, only smaller. What’s really impressive is that you have 2/3s the input and bus capability, 3/4 the matrix with a significantly smaller footprint, all for about 1/2 the price of the SD-8. A standard SD-9 with a 32×8 D-Rack lists for about $27,000. Add a second, redundant power supply for another $2,000.

Yes, the SD-9 is a little more expensive than an M7 or an SC-48. However, with the super-flexible bus and surface structure, and arguably better sound, I think it’s a good value. In the end, it comes down to what is important to you. If I were spec’ing a church install with a console budget of $25K today, I would certainly have the SD-9 on my list of boards to consider. Having spent considerable time on the M7, and in the Avid software (though not nearly as much on the surfaces), I would find myself leaning toward the SD-9. I have already become spoiled by the flexible bus structure, stereo channels, the surface configuration and the sound of the DIGiCo consoles. In just a few short weeks, I’ve become completely at home on the SD-8. Going to the SD-9 was not a big leap, and only required a few changes in workflow.

For a volunteer sound team, deciding between an SD-9 and an M7 will come down to what you value. If having every fader on the surface is of the utmost importance (because the goal of the team is to simply manage volume), the M7 is a good choice. If you are mixing at a higher level, and your team understands EQ, compression, gating and the proper use of FX, I think the SD-9 is quite a bit faster to get around. I also greatly prefer the SD-9’s show-based file system with snapshots compared to the M7’s scene-based arrangement. Arguably, the SD-9 takes a bit more time to set up, configure and get ready to mix. Once you spend that time with it however, it’s an absolute joy. DIGiCo has been very good about releasing software updates with fixes and enhancements, and I expect the SD-9 to continue to grow and develop as time goes on. It’s certainly one you should consider if you’re in the market for a mid-priced digital console.