Friday, December 07, 2012
Midas PRO2 Heads Up System Tailored For The Live Oak Music Hall & Lounge In Ft. Worth
Console serves a wide variety of artists appearing at new live performance venue
Having spent 38 years in the construction business, Bill Smith of Fort Worth, TX decided the time had come to build something for himself and the community.
The result is The Live Oak Music Hall & Lounge on Ft. Worth’s near south side, an architecturally historic Lions Club that Smith gutted and redesigned.
“I’ve always been involved in live music, but never owned a venue before,” he says. “I wanted to create a sophisticated adult listening room that sounds great and where artists want to play. That was the basic idea behind The Live Oak.”
Smith worked with Chris Jordan of Electro Acoustics in Ft. Worth on all the performance aspects of the design. Electro Acoustics is a design/build firm that employs a comprehensive approach to audio, video, lighting and acoustics.
“This was a great project because Bill Smith had a clear vision and trusted us to make it happen,” says Jordan, the firm’s president and chief steward. “It’s an intimate space, but had to be good enough to attract national acts, both in terms of the room itself and the rider-friendliness of the sound system.
“Originally, we didn’t think we could afford a Midas console, but then the PRO2 came out. When we saw the demo, we just instantly fell in love with it. It’s the best digital console we’ve ever seen in that price range, by far. And obviously, the Midas name makes it extremely artist-friendly. It was a perfect fit for us.”
Beyond the actual price, The Live Oak and Electro Acoustics found the PRO2 saved money in other ways. The console’s ample EQ and effects engines eliminated the need for outboard gear, while the Cat-5 fiber connectivity meant no bulky copper snakes, even with the DL251 I/O box located onstage.
Jordan notes, “Not only do we save significant cash in terms of hardware and wiring, but also in terms of space. The PRO2 lets us run both the house and monitor mixes from one surface, but still keep the control booth small enough to give about five extra seats, which was a big consideration for an intimate venue like The Live Oak.”
Another consideration was ease of use. “It’s really a very intuitive board,” says Jordan. “I basically taught it to myself on a Saturday afternoon. Once I figured out the patching, the rest of it really began to flow. I love the ergonomics, especially the POP Groups, and the instant access you have when making adjustments. You’re always just one button away from anywhere you need to go, so the navigation is incredibly fast, which is really important in a setup like this. It’s really the most fun board to mix on that I’ve ever seen.”
With room for about 200 patrons seated or 500 standing, The Live Oak features both local and national artists, with the added attractions of a high-end restaurant and a bar featuring over 200 beers.
In addition to the main music hall, the venue also boasts a lounge with live music, plus both street-level and rooftop outdoor patios. Since its June opening, the venue has hosted such artists as John Oates (of Hall and Oates), Iris Dement, The Vespers, and Jim Lauderdale.
“We’ve been getting rave reviews on the sound of the room, and the Midas PRO2 is a big part of that,” says Smith. “Visiting engineers love it, and they generally need less than 30 minutes to feel comfortable on the console. Our first concert was John Oates. He raved about the room and said he wants to come back.
“In fact, we keep a journal in our green room for artist comments, and everyone who’s played here has basically said the same thing. That’s something we want to build on, to earn a reputation as a sophisticated, top class listening room where artists want to play.”
The Music Group
Not So Mysterious: Using Polarity As A Tool For Optimizing Drum Sound
A method for quickly finding a consistent starting point can help take some of the voodoo out of your system
It’s pretty common knowledge that if you get the wires mixed up when hooking up two loudspeakers that something “not good” happens.
Loudspeaker phase (actually, polarity) seems at first glance a pretty simple concept. If both loudspeakers are moving outward at the same time the sound adds together, and if one is moving out while the other moves in, the sound cancels out, especially the low frequencies.
Hearing this effect is quite easily demonstrated - listen to your home stereo loudspeakers while standing midway between them and then listen again after reversing the leads on one side. You should notice a very apparent decrease in the lows when they’re wired the wrong way.
To picture why this happens, imagine a very simple pulse or “positive pressure wave” being reproduced by both loudspeakers simultaneously. The two positive pressure waves add together and that means addition or “louder.”
Reverse the leads to one of the loudspeakers and one loudspeaker moves outward (toward you, positive pressure) and the other moves inward (away from you, negative pressure).
The pressure wave from one loudspeaker is being “sucked out” by the other loudspeaker, also know as cancellation or “not as loud.”
If we reverse the other loudspeaker lead as well, so that both loudspeakers have reversed leads, both will now move away from you. The two negative pressure waves add together, and that is once again addition or “louder.”
In most situations, it doesn’t much matter whether both loudspeakers move toward or away from you, as long as both are doing the same thing at the same time.
If it was just loudspeakers, all this would be easy .But things are rarely ever simple in the real world of live audio. There are microphones, amplifiers, instruments, drums and plenty more that also make lots of noise. Add in monitors pointing in various directions, and some interesting things happen.
Everyone Does It (Just About)
Nearly every engineer that uses a snare bottom microphone naturally reverses its polarity. Seems simple enough, and you can hear the added lows and punch when pushing the button next to the word “phase” on the console when both the top and bottom snare mic channels are at similar levels.
Fair enough - drummer hits snare, it’s head moves down/away from the top mic while also moving down/toward the bottom mic. The top mic sees a negative pressure wave and the bottom mic sees a positive pressure wave.
It’s not too much different than our loudspeaker example - negative pressure wave plus positive pressure wave equals cancellation, and for most applications, cancellation equals bad.
The simple solution is to reverse the polarity of either the top or bottom mic to create addition instead of cancellation.
The Way Things Are Supposed To Be
First let me state that a positive pressure on a “pin 2 hot” mic should produce a positive voltage on pin 2 of its XLR output connector.
Assuming you have a properly wired pin 2 hot sound system, that positive voltage on pin 2 will eventually manifest itself as a positive (outward) motion on the loudspeakers in your system. (Yes, I know some JBL stuff will go inward, and there are other exceptions to take into account.)
But why does this matter beyond snare bottom?
Because if you mic the kick from inside the drum (non beater side of the head), then the kick drum beat will produce a positive pressure on it’s mic, and therefore, in most pin 2 hot systems, a positive outward motion of the drum monitor loudspeakers.
Imagine sitting at the drum kit: hit the kick, the head is moving away from you, the drum monitor loudspeaker is moving toward you (negative pressure wave from the kick and positive pressure wave from the drum fill). What do we have? Cancellation, just like in the home stereo with one loudspeaker reversed.
And not only do you have cancellation, but there is another problem as well; the drum head moves away from the drum monitor, the drum monitor loudspeaker moves toward the drum head and pushes the head in a bit farther, and then the reverse happens - the drum head rebounds toward you, while the monitor loudspeaker moves away.
The drum head and monitor loudspeaker are augmenting each others’ motion, creating “resonance” and increasing feedback susceptibility. This is generally not a good thing.
But all you need to do is reverse the kick drum mic polarity, and now the kick drum and drum fill outputs not only add together, but the combination becomes overall less resonant.
Makes Sense… But Does It Really Matter?
If you try this approach, keep in mind that time delay induced by digital processors and drum fill placement can greatly affect audibility.
So to really hear the difference and set the proper polarity, bypass all drum fill and kick channel EQ, and then bring up the kick in the drum fill to the point where it is just on the verge of feedback.
Now, by pressing the phase reverse on the kick channel, you should be able to determine which way is noticeably more stable.
OK, Did That. Now What?
If you did this right and have found a clearly better sounding kick polarity, use it as a starting point to set polarity for the rest of the drum kit.
Working around the rest of the kit, for every mic for the under side of a drum, set the phase the same as you did for the kick. And for every mic for the top or “stick side” of a drum, set the polarity the opposite of how the kick is set.
Keep in mind that this is a starting point - upward facing monitors near a floor tom tend to be more stable with the opposite polarity than this approach dictates.
Polarity is another tool that you can use to achieve the sound you’re seeking. here is no “right” or “wrong” way but having a method for quickly finding a consistent starting point can help take some of the voodoo out of your system.
What About The Cymbals?
Polarity issues are going to have the most noticeable effect on low frequencies, as well as when two sources are reproducing similar signal are in very close proximity to each other or mixed together.
Higher frequencies and things panned hard left and right in the mains will often sound different, but there will not be a definitive “ better” sounding choice.
How far is too far? I personally take it pretty far, and further, follow this same pattern on my house console as well. Both my kick and snare bottom are reversed polarity (I top mic my toms).
But how much of this you employ, is, of course, up to you. What we’re doing here is establishing a clean stating point for you to make decisions to change polarity on specific channels to achieve desired results.
Dave Rat is the co-founder and owner of Rat Sound, a leading sound reinforcement company based in Southern California.
Thursday, December 06, 2012
Church Sound: Mixing Console Management
Tried and true methods for really helping yourself out in a live setting
It’s typical to find articles suggesting preparations for many of the elements outside the mixing console prior to worship. After all, we know those things can help ensure an efficient and confident atmosphere during your setup and rehearsals.
However, I wanted to tunnel down a bit further and deal specifically with configuring your mixing console to put you in the best position possible to mix the event and generally make it easier to be a little “quicker on your feet.”
There are few tried and true methods for really helping yourself out in a live setting, where things can change at the drop of a hat and leave you scrambling to recover.
One of the stresses that comes with any suddenly changing situation is the ability to think clearly and logically while you’re scrambling to recover.
Certainly one of the most basic methods for combating the confusion factor is to layout your inputs and outputs on your console in a logical and effective manner.
The first thing I’m going to suggest is, on the surface, very simple, but may require a little thought and creative patching on your part to implement.
In my opinion, this is a very important method to grasp - especially if you’re contemplating moving to a digital console where an input can be plugged in, and then show up, on any fader you desire by using the digital patch bay.
In fact, in some cases with digital, the fader’s name, source, or even position on the console can change during a scene recall.
So with that in mind, I make every attempt to layout my console inputs just as I see them on the stage.
Here are some examples: When I layout my drum kit it will usually go something like this-kick, snare, floor 2, floor 1, rack 1, ride cymbal, overhead left, overhead right, hi hat. This makes tracking down things like a given tom or an overhead channel easy because you can correlate them to what you actually see when you look up at the stage.
It also makes any panning very intuitive. For example, the left-most tom fader would be panned the most left, the right-most tom panned the most right. All cymbal inputs reside next to each other, and you can easily see the relative panning and fader levels of the cymbal mix.
This concept is especially helpful with a lot of players and a lot of instruments on stage, such as a number of vocalists.
Let’s say I have a main vocalist stage center, but I also have four vocalists to either side of the singer It’s well worth the trouble to lay out these inputs to reflect their placement on stage. So, from my perspective, the inputs would layout BV 1, BV 2, lead vocal, BV 3, BV 4.
Also, in this case, I might even try to give myself some hints at what kind of parts the vocalist are responsible for-bass, bass, lead, tenor, soprano. The idea is to allow you to attack these inputs quickly when needed, without having to search for them, especially for console setups with a large number of inputs.
The same principle applies for keyboards, horns, and guitars, especially when there is more than one player involved.
Lay them out on your console in a way that is coherent to what you see when you look at the stage, and your ability to anticipate, react, and even create relative blends will improve dramatically.
Once you have your inputs laid out the way you want them, you can move to the next stage of your setup, which is employing grouping.
For my money, grouping is one of the key areas of a console setup that often pays huge dividends while mixing an event.
Good grouping practices are key to getting a mix together quickly and, at the same time, allowing you to make refined and controlled movements during mixing.
Two Key Grouping Styles
While there are many new and exciting styles of groups coming online with the outbreak of digital live sound consoles, for the purposes of this article, I’m only going to concentrate on two styles: “audio sub groups” and “VCA groups.”
Given that the vast majority of consoles, analog or digital, offer these two styles of groups, I strongly encourage you to thoroughly understand the differences between them and work toward using them.
While many mix engineers tend to use one or the other, they are certainly not mutually exclusive of one another, and when used correctly and together, are a very powerful tool.
Let’s start with audio sub groups. Audio sub groups are generally either mono or stereo and, by definition, provide a summing point for a given number of inputs before they then head off to the left/right master output.
This means that any number of audio inputs can be directed through the audio sub group and the group as a whole can then be moved up or down in volume. By soloing an audio sub group, and listening in headphones, you can then monitor the fader balances of all inputs that are feeding the group, including their pan position.
For example, with the push of the group solo button on a drums group you could listen to the relative blend of all the drum mics and, in turn, affect the overall level of the drum kit in the PA system by moving the group fader without having to change the input fader positions.
The input faders would still be feeding any post-fader aux busses even though the audio sub group fader would be at zero.
Additionally, because audio is actually passing through the group, it will usually offer an insert point where you can patch in equalizers or compressors and limiters which, of course, would affect the drum mix as a whole in the PA system.
This is where the difference between audio sub groups and VCA groups comes to light.
A Voltage Controlled Amplifier (VCA) does not offer an actual audio path for the inputs assigned to it. Instead, once a number of inputs are assigned to the VCA fader, it essentially works as remote control of the assigned faders.
For example, if you had a blend of eight input faders and you assigned them all to a VCA group, once you move the VCA group fader down, it is exactly as if you simply reached over and pulled the actual input faders down.
The relative levels between the faders would remain the same, but the levels to any post-fader aux buses would now change by how far down in level you moved the VCA master.
VCAs, generally speaking, do not allow you to solo the group unless it is a destructive style “solo in place” because of the lack of audio passing through the group.
Likewise, it does not offer you the ability to insert external processing on the group as a whole.
So, with these concepts now in mind-I’m recommending the following to those of you who have both audio sub groups and VCA groups on your console. Use them both. But use them for different tasks.
Start by using your audio sub groups to assemble the components of your event mix into groups. For example, 1-drums-loops & percussion, 2-bass, 3-keys, 4-guitars, 5-backing vocals, 6-lead vocals, 7-pastor, 8-media.
Once done, then assign these groups to the left/right master output. Try to stay disciplined and keep like inputs in their respective groups.
Say you have a reverb unit dedicated to the drums - assign the reverb return faders to the drums group. This allows you to listen to the actual blend of the drum inputs against the reverb return while soloing the drums audio group.
Likewise, if you mute the drums audio group, you’ll no longer hear the reverb return, even though the drum inputs are still feeding it.
Now all of your VCAs are available for doing what I like to think of as “focused” mixing. Now you can assign VCAs to inputs that you need access to for any given segment of your event. They’re located in one position and available for immediate level manipulation.
Maybe you have a VCA that is simply assigned to only the kick and snare or just the cymbals, maybe even just the toms. Any of these allows you to accentuate a given fill or breakdown in a song with the movement of one single fader.
Or maybe you have the percussion assigned to its own VCA, with those inputs living as a part of the drums audio sub group. It all just depends on what you need to get to at any given time.
This is a wonderful workflow for digital consoles and even some analog consoles, in that you can program the VCA assignments dependent upon what you need to get to at any given time.
It’s all up to your imagination and, if done properly, there is rarely an excuse for missing cues because you were late finding the fader.
One of the bonus benefits of VCA grouping is that it can be used to control either input or output faders. I like to assign a “band” VCA and a “vocals” VCA - but assign them from the group path, not the input path.
This means you would assigning all of the audio groups that carry music components to one VCA fader and assign all audio groups that carry vocal components to a different VCA fader.
Now, with the adjustment of two VCA faders, you can balance the band mix relative to the vocal mix. This is wonderful for quickly placing vocals at any level in relation to the band mix.
Try using these techniques and you’ll soon be presenting very reliable, controlled mixes to your listeners.
Tuesday, December 04, 2012
Ample Options: The Universe Of Live Recording
Let’s take a look at some of the ways to record live events
There are many “sub-groups” among pro audio folks, with two of the primary groups being live and recording.
But live mix engineers are now being tasked with interfacing recording systems and recording shows directly, whereas in the past we were usually only asked to supply an analog split off the snake for a separate recording company.
With recording increasingly becoming a regular part of our job description, let’s take a look at some of the ways to record live events, along with some of the methods I use for each.
Note, however, that my approaches are by no means the only ways to go about it.
Stereo or 2-track recording (also sometimes called room recording) involves placing two microphones where they can capture the entire sound of the performance (usually in the audience area) and a simple recording deck.
For some shows, such as a symphony in a performing arts center, this may be the best way to capture the music, as the recording can accurately reflect the audience listening experience complete with the acoustic environment.
However, it may not produce the desired results in situations where room reflections and audience noises may mask parts of the program.
When room recording, I usually place two directional microphones in an X/Y configuration (the fronts of the microphones about 90 degrees to each other) on a tall stand. These are located in the seating area, well above the heads of the audience, so they’ll pick up less noise and more program material, as well as some of the natural room ambiance.
During rehearsal or sound check, I put on a set of headphones and listen as I move the mics around to find the best spot. Omnidirectional mics - used singly - can also be a good choice.
On The Board
In live sound, we frequently see a variation of room recording called taping, usually done by fans or associates of the band. (The term, of course, harkens back to the pre-digital era, when recording was done with magnetic tape). Tapers might use separate mics on stands running into small mixers and recording decks, or some of the newer recorders that contain mics, a mixer, and file storage in one compact package.
Diagram of a standard X/Y mic configuration, and inset, the Audio-Technica AT2022, with has two unidirectional condenser capsules in an X/Y configuration pivot. (click to enlarge)
Normally, we’re not involved in assisting tapers, but sometimes artists may ask that a board feed be available for those who wish to record. Note that it’s not a good idea to plug your house mix console directly into an unknown recording rig because it might introduce noise issues.
I use a press multi box (“press mult”) and provide a number of transformer isolated outputs derived from the master bus, matrix or aux outputs. If you’re using an output other than the mains, be sure to feed all channels into the recording send.
Another common technique is the board tape, which is simply taking a copy of the main left and right outputs and running them to a recording deck.
While it’s easy, there are also downsides, primarily because it’s the live house mix that is being recorded.
For example, instruments may not sound “right” because they were equalized to be heard through a PA in a particular room, and not for a recording.
And a lack of audience response on the recording might make it seem like the band was playing to an empty house.
Even with these drawbacks, a board tape can be a useful reference. Artists can get a good feel for how they performed and engineers can focus in on things they did – and did not – do well with the mix.
Many consoles now come equipped with a set of recording outputs, typically RCA connectors, making it a simple affair to interface a recording deck to the console. Some consoles also have dual outputs on the master section that allow a recording feed to be easily patched from the main outs.
A note of caution: do not use the console’s control room outputs for a recording because if any of the pre-fade listen (PFL) or solo buttons are engaged, because this will route only the solo’d signal to these outputs, ruining the recording. (Unfortunately I’m speaking from experience on this one.)
Making It Safe
Board taping has also long been deployed at corporate events, made for legal purposes such as the recording of a stockholder meeting as well as for other general purposes. Often these recordings are duplicated and sold to those who want to hear the information but were unable to attend the seminar directly.
While numerous companies specialize in the recording of corporate events, and our involvement with them as a tech or engineer may be to simply provide a single board feed, it’s not uncommon at smaller events for us to be asked (many times at the last minute) to record a speech or assembly.
For general sessions at larger corporate events there may be the need for a safety - as the name implies, this is a copy made just in case something were to happen to the main recording, or video audio of the event. It might be recorded on tape, CD, hard drive, or even sometimes on the audio tracks on a spare video deck using video media.
A typical rig deployed by the author for corporate events, and it includes an A/D converter feeding a laptop PC for live recording. (click to enlarge)
I’ve heard of more than one instance where the safety saved the day when a video deck malfunctioned or the package containing the primary recording was lost in transit.
One technique I use for a safety recording is to place a mic splitter on the podium mic or lavalier mic of the presenter, and then run a split directly into a recorder, bypassing the console completely. This may be the only way to record the presentation if you’re using a small console with a limited number of outputs.
A variation on the board tape is a mixed board tape, where a separate mix of the program material is done specifically for a recording. Sometimes this can be done by simply using the matrix section of the console, feeding it the mix and any additional components (such as the instruments noted earlier) that are missing in the PA mix.
It can also be as complex as running a separate aux send mix for every input and mixing an entire separate program for the recording. This is very popular with bands who want a live recording on a budget, or who want to provide their fans with a live recording right after a show.
I’ve found that monitor desks can be perfect for mixed recordings if the extra outputs are available. And depending on the band’s monitoring requirements, the monitor mixer may also have more time available than the front of house mixer to attend to the recording.
As noted earlier, recording splits from a snake system have been a very common way to interface a live recording rig into a system.
Analog snakes set up for recording should contain transformer isolated outputs, and sometimes ground lifts on every channel, so there will be no noise issues between the live and recording setups.
Remote recording companies can simply tie their lines into the isolated splits and have access to all the stage signals. If no snake split exists, direct outputs from a console can be used to feed a multi-track recording system.
Digital snake systems and networks present an advantage over analog systems when it comes to recording. Instead of having just one isolated output located at the snake head, many digital transport systems have multiple splits that can be placed anywhere along the network, accommodating remote recording, webcasts, broadcast feeds, and any other sends that may be required.
The drawback is that in many cases, the digital signals may need to be converted into analog for the recording gear.
Depending on the transport system protocol, remote recording setups might include digital mixing consoles or digital recorders that can interface directly with the digital snakes, eliminating D/A converters in the signal chain and making hookup as simple as running a length of Cat-5, coaxial cable, or fiber optic cable to the recording console or deck.
Digital mixing consoles bring a whole new wealth of capabilities for recording. Many are designed to interface directly with computer based multi-track recording systems.
Whether configured to do so from the factory, or by using optional output cards, some systems can output MADI, a digital protocol with 64 channels of audio, or ADAT optical, a digital protocol that sends 8 channels down each optical cable.
Both of these are used by many recording systems for multi-channel audio transport. AES/EBU and S/PDIF are two other common digital audio protocols used by many manufacturers to interface recording and playback equipment.
In some cases, a separate recording deck may not even be required, as a few digital consoles have the capability to record via USB to a memory stick or optional hard drives. These files can be downloaded after the show to computer editing programs.
New trends in the market are analog consoles that output audio to FireWire or USB, allowing connection of the console to a computer. These consoles can usually do double duty acting as a live mixer at a show while simultaneously sending audio to recording software in the computer.
Many of the smaller ones are tailor-made for corporate meetings where limited channel counts and the ability to record sessions are the norm, or for engineers and sound companies seeking to get a lot of bang for the buck in gear purchases.
With the ability to interface multitrack recording units easily into live rigs, one new concept that has taken hold is the virtual sound check. Simply put, this is where a multi-track recording is made of the act, and each track is played back into the same inputs. It puts the “band in a box,” so to speak, allowing engineers to do things at sound check like tweaking effects, programming scenes, or checking how the room sounds without needing the musicians on hand.
While used by many on gigs, it’s also a great way to train less experienced/aspiring audio folks, allowing them to spend hours learning mixing techniques and refining their skills without making mistakes at a show. This also comes in handy for veteran mixers who want to become familiar with a new console without the pressures of a real gig.
Live Recording Checklist
No matter how a recording is achieved, certain things should always be done to insure the best results.
—Start with good mics optimized for the recording. While many live mics can make good recordings, some are a compromise between ruggedness and sound quality. Studio mics may be a better choice for certain applications.
—Mic placement onstage is critical. You want to minimize bleed from other instruments as much as possible into each mic. While this may not be a big deal on a simple board tape, it’s critical for making good multi-track recordings.
—You may also have to move a few pieces of backline around, or isolate loud instruments. Plexiglass shields around loud instruments like drum sets keep the drum sounds out of other microphones, as well as limit the amount of guitar/bass amplifier sounds bleeding into the drum mics. Packing blankets between or behind loud amps can help control spill.
—Record tracks with the EQ flat, and with no effects or compression. EQ and effects can always be added in later during the recording mix-down.
—If possible, build in additional time for set up and sound check. Mic placement might take longer, and it’s also good to have the chance to do a test recording listen to the playback to ensure that everything is OK.
—Monitor the recording throughout the performance. Check levels, and verify all the gear is running correctly.
—Check the recording times for the media used and allow ample time to switch flash cards and CDs, or to flip or exchange tapes. Plan ahead and map out points in the show where you will have a moment to accomplish the switch without missing any of the program.
—Label all recordings with the name, date and any other notable specifics, and be sure to do this to both the media and the case to keep it better organized.
Craig Leerman is senior contributing editor for Live Sound International and ProSoundWeb, and is the owner of Tech Works, a production company based in Las Vegas.
Sunday, December 02, 2012
How To Tame The DAW Live Recording Monster
Using a DAW for live recording isn’t terribly difficult as long as you give yourself sufficient time to set it up.
There are pros and cons in using a computer-based DAW (digital audio workstation) to make a live recording.
While on the surface that seems like a no-brainer, a lot of engineers are afraid enough about the setup time and the supposed fragility of DAW in a live situation to feel comfortable with them.
When DAW platforms like Cubase, Pro Tools, and Sound Forge (to name just a few) first came on the recording scene in the mid-1990s, you might say that their initial use for recording was a bit on the experimental side.
This was O.K. for the studio, where although you might be paying for the time, at least there wasn’t the life or death sword of a live show hanging over your head.
Not only that, the software was sometimes not that intuitive and just took too long to get a handle on, so the whole concept of a casual DAW recording was totally dismissed. Better to leave that can of worms to the pros.
A lot of live engineers remember those days and mistakenly believe that the state-of-the-art still remains in that era.
Luckily, the technology has advanced quite a bit from those days, so recording with a DAW is now easier than ever.
Today’s software is rock-solid, the computers are robust, and the interfaces are well-built, each one a far cry from what was the norm 10 years ago.
Even the most inexpensive DAW setup can now provide a great recording, with far more available features than are required for this type of application.
Let’s look at some of the upsides and downsides of recording a show with a DAW.
On The Upside
A DAW just isn’t that expensive anymore. If you already have a laptop, you can be in business for less than 500 bucks. You get what you pay for, of course, with better converters and more inputs and outputs as you move up in price.
Figure 1: Using the insert for a direct output.
Eliminates a transfer
Any multi-track recording no matter how large or small will ultimately end up in some sort of DAW for mixing.
It’s so much faster and easier if it was recorded in the format that it will be mixed in to eliminate a transfer.
Transferring files can sometimes be a little tedious, especially if you have dissimilar formats.
Flexibility during mixing
This is by far the number one factor for recording with a DAW - there is far more flexibility for mixing compared to just about any other format.
On The Downside
Not only do you have to do your normal sound check, but now you must make time for the setup of the DAW and a recording sound check as well.
What computer doesn’t fail, especially during a critical operation?
If there is a power failure or a software error, it can take at least several minutes to get back up and recording, which might be unacceptable for your situation.
Requires knowledge of the software
Sure, most DAW software packages seem easy on the surface, but when you really get into it in the heat of battle, you need at least a little knowledge of the app to make sure everything is routed correctly in the first place, and to allow you to be able to deal with the many small things that inevitably pop up.
Note, however, that these downsides can be overcome. Here’s how:
Schedule a longer sound check
Realize that extra time is needed just to deal with the recording. The time will be divided into setting up the DAW and then for signal routing and level check.
Record some of the sound check
Having recorded audio to play back can help you tweak the setup later, after the sound check ends. Also, you never know when you might capture something great. Sometimes just having a piece of a song to cut in later when mixing can be a big help.
Figure 2: The Metric Halo built-in record panel.
Set levels low
There isn’t much time to check levels during the show, so set levels on the low side so you won’t have to worry about any overloading later.
Don’t do any processing during the recording
Every processor plug-in (like a compressor) will impart a slight latency, which makes it difficult to monitor while recording because of the time delay between the headphones and the live band.
If you don’t have a splitter, use the direct outs
The direct outs are an effective way of getting the signal into a DAW without renting an expensive mic splitter.
Just remember that any processing done on the console will also be done to the signal as well. If the mixer doesn’t have direct outs, use the out side of an insert (Figure 1). The subgroups or a console matrix (if you have one) are also good ways to route the signal into the recorder.
Use a UPS
If using a desktop computer, an Uninterruptible Power Supply (UPS) will ease any strife you might have about the AC power going down. It won’t cover a computer failure, but if power is lost, at least what’s already been recorded can be saved. You won’t have this problem with a laptop as long as you’re running on AC power (which is highly recommended) because the computer will automatically switch to it’s internal battery.
Use the simplest way to record possible
For example, Metric Halo interfaces offer a feature called the Record Panel that is extremely simple to use and eliminates the need to even open up a DAW software application (Figure 2) until it’s time to edit and mix.
Using a DAW really isn’t that difficult these days as long as you have sufficient time to set it up and follow the suggestions presented here. For further routing ideas, check out an earlier article in this series, Multitrack Made Easy: Recording With An 8-Channel Interface.
Bobby Owsinski is a veteran audio professional and the author of several books about live and recorded sound.
Saturday, December 01, 2012
Live Sessions: Multitrack Recording Made Easy
Recording With An 8-Channel Interface.
These days it seems that almost everyone has a digital audio workstation (DAW) and a common 8-channel interface from Avid, MOTU, Edirol or some other manufacturer.
Believe it or not, these small, inexpensive rigs pack more power than The Beatles ever had during their heyday (yet they sold over 1 billion records), and we can easily put that power to work when making a simple 8-track recording.
While many of the interfaces actually have more than 8 inputs if you combine the analog and digital inputs (plus some secondary inputs for aux or 2-track returns), we’re just going to use one set of eight for this illustration to keep things simple.
After all, the idea is to make a multitrack recording that’s easy to set up and as seamless as possible.
Since there are a limited number of inputs, the track assignments for the various instruments and vocals have to be thought out carefully in advance.
Ideally, you want the lead vocal on one track, all of the background vocals (if there are any) on one track, kick and snare on separate tracks, and guitars, bass and keyboards on separate tracks.
Figure 1: Direct Output (above) and Channel Insert
Usually the rest of the drums will be picked up by the stage microphones, especially if the band is on a small stage, so while this isn’t ideal, it does provide a measure of control over the most important elements.
A separate “kit” mic placed a foot or so above the drummers head and aimed at the middle of the kit is a better way to go if it’s possible, but this discussion is focused on making things easy.
How do you get these mix elements separated out? Almost all mixers and consoles at just about any price past a few hundred bucks these days have either Direct Outs or Inserts on each input channel.
Figure 2: Plug inserted halfway into an insert jack
A Direct Out does just as it’s name says, serving as an output from only that isolated channel that’s usually intended for recording. Just plug that into your interface for the lead vocal, kick, snare, and bass.
If there is not a Direct Out, it’s almost certain that there’s an “Insert” on each microphone channel (Figure 1).
As you’re no doubt aware, an Insert jack allows insertion of a device like a compressor, delay or reverb only on that channel, but it can also be used as a direct output. The trick is to push the plug in only halfway (Figure 2).
Variables & Options
What if you have two guitars and two background vocals (or more)?
You have four musical elements to record but only two additional input channels on your interface.
The answer is to use the “Sub-Groups” that you probably also have on your mixer or console (Figure 3).
Route the guitars to a sub-group and plug the output of the group into your interface, then route the background vocals into a second sub-group and send that to another channel of the interface (Figure 4). And that’s it.
To finish the recording, place an audience mic about midway between the stage and back wall of the venue pointing towards the stage.
The audience mic will not only pick up the audience response, but should also give the track a little “glue” by helping to balance everything together with some of the sound off the stage.
Figure 3: Mixer Sub-Group
Don’t add too much though, as a little goes a long way. Don’t forget to align the time of the audience track with the others in the DAW to keep the sound punchy and eliminate any possible phase cancellation (see Live Sessions from the May 2010).
Time To Mix
You’ll want to take a different mix approach for a live recording than for a live mix, and different from a studio recording for that matter.
In the studio, some engineers usually start their mix with the drums (either the kick first, but sometimes the snare or overheads), bass, or vocals. It all depends on the song or the particular preference of the engineer.
In live recording, it’s better to not be as stringent on where you start the mix from, since it could lead to emphasizing the track with the weakest performance. Rather, what you’re trying to do is shift the attention to the strongest track.
Figure 4: 8-channel live recording setup
Even though an 8-track mix is fairly simple, figure out the tracks that are the weakest and don’t add them to the mix until almost last (the audience track will be absolutely last).
If the drums are a little floppy execution-wise, start with the bass and keep that in front (just like in 1960s rock/pop records).
If the vocals go a little “sideways,” keep them lower in the mix. At all times, mix to your strengths, de-emphasize your weaknesses, and bring the audience track in last to add the “glue”.
Next time I’ll discuss the one element that truly makes a live recording “live” - the audience mics.
Bobby Owsinski is a veteran audio professional and the author of several books about live and recorded sound.
Monday, November 26, 2012
In The Studio: Eight Indicators That A Mix Is Finished
You might be finished before you know it
One of the tougher things to decide when your doing a project is when the mix is finished.
If you have a deadline, the decision is quickly made for you, but if you have a deep pocket budget or unlimited time, a mix can drag on forever.
So when is a mix considered finished? Here are some guidelines, courtesy of The Mixing Engineer’s Handbook:
1) The groove of the song is solid. The groove usually comes from the rhythm section, but it might be from an element like a rhythm guitar (like on the Police’s Every Breath You Take) or just the bass by itself, like anything from the Detroit Motown that James Jamerson played on (Marvin Gaye’s What’s Goin’ On or The Four Tops’ Reach Out, I’ll Be There and Bernadette for instance). Whatever element supplies the groove, it has to be emphasized so that the listener can feel it.
2) You can distinctly hear every instrument. Every instrument must have its own frequency range to be heard. Depending upon the arrangement, this is what usually takes the most time during mixing.
3) Every lyric, and every note of every line or solo can be heard. You don’t want a single note buried. It all has to be crystal clear. Use your automation. That’s what it was made for.
4) The mix has punch. The relationship between the bass and drums is in the right proportion and work together well to give the song a solid foundation.
5) The mix has a focal point. What’s the most important element of the song? Make sure it’s obvious to the listener.
6) The mix has contrast. If you have the same amount of the same effect on everything (a trait I hear from so many neophyte mixers), the mix will sound washed out. You have to have contrast between different elements, from dry to wet, to give the mix depth.
7) All noises and glitches are eliminated. This means any count-offs, singer’s breaths that seem out of place or predominate because of vocal compression, amp noise on guitar tracks before and after the guitar is playing, bad sounding edits, and anything else that might take the listener’s attention away from the track.
8) You can play your mix against songs that you love, and it holds up. Perhaps the ultimate test. If you can get your mix in the same ball park as many of your favorites (either things you’ve mixed or from other artists) after you’ve passed the previous seven items, then you’re probably home free.
In the end, it’s best to figure at least a full day per song regardless of whether you’re mixing in the box or on an analog console, although it’s still best to figure a day and a half per mix if you’re mixing in a studio with an analog-style console.
Of course, if you’re mixing every session as you go along recording, then you might be finished before you know it as you just tweak your mix a little.
Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.
Allen & Heath Launches New AES Option For iLive Digital Mixer Series
Provides two AES3 digital 2-channel outputs in place of four of the analog line outputs
Allen & Heath has launched a new AES Out option module for the fixed format MixRacks from its iLive digital mixing series.
Specifically, the iDR-16, iDR-32, iDR-48 and iDR-64 fixed-format MixRack or the xDR-16 expander unit are fitted with an AES digital output option, providing two AES3 digital 2-channel outputs in place of four of the analog line outputs.
Supported by the newly released firmware V1.9, the option is also available as a retro-fit kit for existing customers, allowing single or multiple AES Digital Output options to be fitted.
“A&H is really pleased to be able to offer the option for AES outputs on all of the fixed format MixRacks in the iLive family,” says A&H product manager Leon Phillips. “The launch comes in response to engineers and installers who want to keep the connection to the PA in the digital domain.
“Previously only an option on the flagship modular iLive systems, the new AES options will place fixed format iLive systems in the running for new applications, such as cost-sensitive broadcast installations.”
Allen & Heath
Church Sound: Sometimes It’s The Simple Things That Matter Most
Sweating the small stuff ahead of time can pay off
I can’t believe that I’m writing about such a simple thing.
It’s something that should be a given, a natural for every sound person to do for every gig.
Yet I find novices to seasoned pros not doing it—and then paying the price.
What is this one simple thing? Checking every input before the service or event starts.
By checking every input, I mean either using headphones or turning up each input one at a time and listening to see if the input is working, and further working properly (no noise on the line, etc…).
By the way, this test should also include checking the battery level on all wireless systems.
I’ve made it a habit to have each pastor talk over the sound system right after I give them their wireless mic. This check usually happens one hour before the start of the service while the musicians are rehearsing.
Depending on the pastor, and the trust level we have with each other, I ask them to keep the mic on or to promise that they will turn it back on before the start of the service.
If it’s a guest pastor, I usually turn the belt pack on and then tape over the switch so it can’t be easily turned off.
It’s also best practices to do a “line check” prior to each service/event, listening to every input on headphones using the PFL (pre-fade-listen) on each channel.
Generally, I do this somewhere around five minutes before the start of the service. My logic is that it’s close enough to the start to make me feel comfortable that nothing will go wrong or change. It also leaves me with enough time to correct or change the plan if something is not working properly.
One last thing: if we’re planning on showing a video during the service, I have the video guys run the entire video with audio playback going through the main sound system.
This is done for two reasons. First, so that I’m not surprised by a jump in audio level during the service, and second, to make sure that the video won’t freeze while being played in the service.
As I noted at the beginning: this is all simple stuff! But I bet it’s happened to all of us at least once, if not more. And we’ve all been at events where a person gets up to speak and the mic’s not working.
“Don’t sweat the small stuff” is a common bit of advice, but perhaps it would be more accurate to say, at least for sound operators, “Sweat the small stuff ahead of time so you don’t have to sweat when it counts.”
Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at churches for more than 30 years.
Wednesday, November 21, 2012
Bryony October Takes Soundcraft Si Performer On The Road For Delilah Tour
An early adopter reports on her first outing with the new console
Mix engineer and production manager Bryony October recently took to the road with British singer songwriter Delilah, and at the same time became an early adopter of the new Harman Soundcraft Si Performer digital console that builds on the successful Si platform while also incorporating a DMX lighting interface.
October needed a minimum of 31 inputs in mixing the five-piece band’s monitor in-ears from front of house. But her chosen console also needed a small footprint in order to fit into a splitter van (along with IEM rack and mics).
Knowing that this tour would not warrant full production support, October had earlier contacted Soundcraft to see if they had any new consoles in prospect. “The timing couldn’t have been better as the Si Performer was just being released,” she notes.
The Si Performer’s FaderGlow and color backlit name displays on every channel proved valuable. “It has screens on each channel, so you can see the compressors and gates, and the FaderGlow changes color so you never make a mistake when moving from FOH to monitor mixes,” she explains. “It also turns the system EQ red so you can clearly see the graphic.”
“The touch screen is easy to get around quickly, with the one-touch easy mix with buttons across the top of the faders,” October adds. “Finally, I love the magnetic iPad notes pad which is very handy.”
The entire show mix is placed on the desk’s top layer, with October sub-mixing the drums to a group bus and also Delilah’s vocals and some of the drums to the Lexicon effects.
As for Delilah’s personal monitor mix, she says, “It’s all about the vocal level, reverbs [the desk contains four stereo Lexicon effects engines] and a low mix of everything else. It’s about creating an open and ambient sound so that the artist feels part of the show and can interact with the audience.”
Sonically, October believes, the desk is excellent, and although she hasn’t had the opportunity to use the DMX interface, she notes that “every single venue I arrived at knew that this was the desk with DMX.”
Tuesday, November 20, 2012
Lectrosonics ASPEN Automix Capabilities Benefit Florida Courthouse
Smooth operation of multiple microphone environments
The recently renovated 15th Judicial Circuit of Florida West County Courthouse in Belle Glade needed an AV solution to manage multiple microphones and a plethora of audio, video, and teleconferencing equipment. In addition, court proceedings are commonly recorded.
Managing all those audio feeds required a sophisticated processing system with the ability to interact seamlessly with other equipment—challenges met by Lectrosonics ASPEN processors.
Ft. Lauderdale, FL-based NDR Corporation, a design/build firm providing turnkey AV installations, was contracted to handle the West County Courthouse’s facility upgrade. Specifically, NDR deployed SPN1624 ASPEN processors, SPN16i ASPEN 16-input expansion units, SPNConference ASPEN telephone conferencing processors, and PA8 8-channel power amplifiers.
“This was an involved project,” states NDR sales manager Byard Hey, “The job involved two courtrooms and two hearing rooms. In each of the two courtrooms and one of the hearing rooms, we deployed the same setup, which included the ASPEN SPN1624, the SPN16i input expander, the SPNConference processor, and two PA8 power amps.
“The second hearing room used the same equipment—minus the SPN16i expander, as this space required fewer audio inputs.”
Unattended operation was another primary aspect of this job, as was the ability of the ASPEN processors to operate as an integral part of the managing Crestron control system. In each of the rooms, the judge has a touch panel controller for making changes to various aspects of the system and there is also a wall panel for the bailiff to interact with the equipment if necessary.
“The ASPEN system is set up for automixing,” Hey explains, “with an average of thirteen microphones in each space, as well as audio from portable AV sources such as laptop computers, video conferencing, telephone conferencing, audio recording, and video follow audio for camera initiation. All of these feeds are handled through the Lectrosonics ASPEN family of products.
“I’ve been very impressed with the ASPEN equipment, particularly the automixing process with mix-minus functionality, which is applied for each ceiling speaker associated with an open microphone in close proximity. Equally important on this project was the system’s expansion capability, as this was what enabled us to accommodate all the mic inputs and the additional AV and conferencing audio inputs.”
While configuring and controlling the ASPEN processing system is quite intuitive, Lectrosonics does offer training to help integrators get up to speed with the equipment. Hey took advantage of that training.
“I attended a one week class at Lectrosonics’ headquarters in preparation for this project,” says Hey, “and my lead technician had experience with other Lectrosonics equipment. Throughout the project and during final commissioning, my tech spoke with Lectrosonics’ support staff and, in each case, had a very positive experience.
“We’ve always had a great relationship with Lectrosonics and these factors explain our continued and repeat purchases of their equipment. Lectrosonics provides the right products at competitive pricing with responsive support services—and this helps NDR Corporation protect the reputation we have worked so hard to attain.”
Radial Introduces Submix 4 x 1 Mixer Module For 500 Series
Latest in the company's growing family of 500 Series module
Radial Engineering has introduced the Submix 4x1 mixer, the latest in the company’s growing family of 500 Series modules.
The ergonomically simple design begins with four front-mounted 1/4-inch input connectors with individual level controls.
Unlike a traditional mixer that requires a separate input pad and level control, the Submix employs Radial’s unique dual-gang Accustate input circuit that allows adjustment of sensitivity with gain. This enables the Submix to be used with low level instruments and high output line level sources without the need for a level reducing pad.
The mixer itself follows the same virtual-earth mix bus design as pioneered by classic console makers such as Neve and SSL, so the Submix can be cascaded without adding noise to create larger 8 x 1, 12 x 1 or even 40 x 1 mix formats—depending on the number of modules in play and the Workhorse power rack being used.
This, for instance, enables a 3-slot Radial Powerstrip to be converted into a 12 x 1 keyboard mixer for live use. Using two Submix modules side by side opens the door for stereo applications such as tape returns, digital samplers, stereo effects and analog summing.
As with all Radial products, the Submix is made in Canada using solid steel and finished in a baked enamel. The added shielding of the fully enclosed design helps reduce noise from outside electro-magnetic pollution.
Inside, steel potentiometers connect to a double sided military grade PCB with an RF resistant full-surface ground plane. Double sided gold contacts assure a positive connection inside the rack.
According to Radial sales manager Roc Bubel, “Since our initial foray into the 500 Series format, Radial has strategically developed the most comprehensive series of power-racks on the market. Another aspect to our long term plan is to produce the most expansive and interesting assortment of modules possible. Together we believe that these tools can open up creativity by providing the audio engineer with more options and less boundaries.
“In many ways, the 500 Series format parallels old-school synthesizers where patching between modules can lead to interesting results. The new Submix lets you patch in any four signals and then send the combined output to some other device. Who knows where this will take folks… At the end of the day, it’s all about unleashing creativity.”
The Radial Submix 4x1 module is backwards compatible with older 500 Series formats. It is now shipping, with an estimated retail price of $350 (USD).
Monday, November 19, 2012
Church Sound: Is Your Worship Leader Too Loud?
Keep in mind three criteria when setting their volume level
Dare I suggest the volume of your worship leader could be detrimental to the worship environment?
Yes, yes I suggest that very thing.
Overall audio volume level discussions are common between sound techs but I submit to you, my friends, that the overall volume isn’t nearly as much of a deal-breaker, mood-killer, worship-ender, as the volume level of the person leading the song.
My wife is a wonderful singer (of course!) and has spent a good amount of time singing on a worship team or two.
She also has a good ear for what is and what isn’t a good worship environment. Regarding worship environments, one of the most useful comments she ever said to me was, “when the worship leader’s vocal volume is so high that people can’t hear themselves sing, they won’t sing.”
Let’s dig deeper into that statement…
“When the worship leader’s vocal volume is so high that people can’t hear themselves sing, they won’t sing.”
First, she isn’t saying that we have to hear ourselves sing because we want to hear our own voices. It’s much more of a psychological issue. And let’s make this psychological issue one in which you might be able to relate.
Place yourself in the middle of the congregation during the worship service. Next, imagine the worship leader’s volume is pretty high. You want to praise and so you start singing. You will naturally, without thinking, start singing at a higher volume level so you feel your voice is present as part of the body.
When your vocal volume has to exceed your brain’s “internally acceptable personal volume level” then you start feeling self-conscience.Then you lower your voice. You might even stop singing.
The church sanctuary is a place where you and I should feel free to raise our voices as loud as we want. But for many people, the human psyche places a limit on what is acceptable.
You might have grown up in a church where everyone sang loud. You might have grown up in a reserved church where you softly sang hymns and were looked down upon if you sang out louder than any others.
We should want to lift our voices but the honest truth is that most people have a volume level which, when they cross, they feel self-conscience and their spirit of worship fades away.
Setting The Worship Leader’s Volume Level
You might only have the worship leader as the lead singer or it might vary from person to person. No matter who it is, you need to find a way to set a proper volume level.
There are three criteria I use when setting the lead singer’s volume level;
1) Out-front on new songs. Any time the worship team is singing a new song, the congregation needs to clearly hear the lead singer.
This isn’t to say the lead singer’s vocal line covers up everything else. It’s like walking through a crowded shopping mall and following someone else. They don’t have to be 20-feet tall but they need to be tall enough that you can easily follow wherever they lead you.
2) Lower on known songs. The church body should feel like they are collectively lifting their voice. Let’s stick with the crowded shopping mall analog. You don’t need worry about the location of the leader because you already know where you are going and how to get there.
But they are the leader, so if they divert into an unexpected shop, like the “Sing-the-Chorus-Once-More Store,” then you can still follow along.
3) Always rising above. This would be the “you’ll know it when you hear it” criteria. One of my favorite worship CDs is Yahweh (Live) by Hillsong.
It’s a live CD but I can hear the audience singing along. The lead singer’s voice rises above the unified voice of the congregation but it’s still a part of the overall sound.
Imagine worshiping in song, as being one with the congregation and the band. Imagine then that you stop singing and listen. You hear the voice of the congregation but just above that is the lead vocals.
The Take Away
The volume level of a singer can make or break a worship environment. You should keep in mind three criteria when setting their volume level; louder on new songs, lower on known songs, and always rising above.
If you have a good relationship with your worship leader, then take a page from my playbook: recommend new songs get a “special music” treatment wherein the band plays and the words are displayed on the screen, but the congregation isn’t asked to follow along.
Using this process, the first time the congregation is asked to sing along, they are comfortable with the flow of the song and much more likely to focus on worship rather than focusing on singing the melody the right way.
Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.
Friday, November 16, 2012
Yamaha Commercial Audio Training Sessions Set For November & December
Coming to Washington DC, Michigan, Ohio and Florida
Yamaha Commercial Audio Training Seminars (YCATS) will make stops in Washington, DC; Cincinnati, OH; Novi, MI; and Orlando, FL during the months of November and December, 2012.
YCATS trainers will hit the District of Columbia on November 28-30 (Wednesday through Friday) for Yamaha CL digital console operational training, M7CL digital console for beginners, M7CL for advanced users, digital audio networks for engineers, and an LS9 digital console workshop.
Further, Yamaha CL operational training will be held in Cincinnati on Tuesday, December 4 and in Novi on Friday, December 7.
In addition, Orlando training will be held December 11-13 (Tuesday through Thursday), and will consist of M7CL for beginners, M7CL for advanced users, CL Series essentials (for those who are familiar with “Centralogic” on M7CL or LS9), and digital audio networks for engineers.
Go here for more information and to register.
Yamaha Commercial Audio
UBS Takes First New Soundcraft Si Performer Console In The Baltics
Latvian company takes possession of first digital audio mixer with integrated DMX for lighting control
UBS (Universal Baltic Sound) of Latvia has purchased the first new Harman Professional Soundcraft Si Performer console in the Baltic region.
The rental company had originally been invited by Soundcraft’s Latvian distributor Audio AE to test the Si Compact 24 console on the recent tour of top local rock act Brainstorm, as the front of house mixing console for support band The Sound Poets.
After using it on various shows, they were impressed with the feature set. Although USB senior sound engineer Maris Liepa had never previously worked on the desk, he was surprised at how easy it was to navigate.
“I instinctively knew how to use it without referring to the manual,” he says. “The tour was a great test for a new product and we were able to run both FOH and monitors from a single Si24.”
They were all set to purchase the desk and add it to their extensive inventory of Soundcraft analog boards, until the Si Performer hybrid desk, which is based on the same platform, was announced.
“When we learned of the Performer 3, we were particularly impressed with the bigger channel count, the two option slots, which allow I/O expansion…and the more attractive appearance,” says Maris. “And so we didn’t hesitate to place the order for this instead.”
The Si Performer is the first digital audio mixer with integrated DMX for lighting control. With a total possible capacity of 80 inputs, it takes the Si Compact platform as its base but provides almost twice as much DSP power and increased functionality.
The Si Performer 3 offers 32 mic and eight line inputs, offering 14 aux/group mix buses, as well as four FX return channels; and with the aforementioned input cards it is possible to patch up to 80 inputs.
In addition, the integration of a DMX512 port offers core lighting control, with the initial release of software providing four scene masters (A-D) with associated slave channels on the ALT fader layers.
Posted by Keith Clark on 11/16 at 09:48 AM