Monday, August 08, 2016
Ferrofish A32 Converter & RME Interface Play Handy Role For Ellie Goulding On Tour
New Ferrofish A32 AD/DA converter for the keyboard and playback rig, joined by RME MADIface XT 394-Channel, 192 kHz USB 3.0 audio interface
On the last leg of Ellie Goulding’s recent tour of Europe, next coming to North America, the sound team upgraded the keyboard and playback rigs with technology from Ferrofish and RME, which are both distributed by Synthax.
Specifically, Will Sanderson, the tour’s MIDI, playback and keyboards technician, decided to deploy the new Ferrofish A32 AD/DA converter for the show’s keyboard and playback rig and the RME MADIface XT 394-Channel, 192 kHz USB 3.0 audio interface.
“The aim was to design a system that would handle keyboard sounds, electronic drum sounds, playback, plus autocue and timecode while enabling the musical director and the musicians to have free reign with their sound design and full control of it during the show from on stage,” Sanderson explains. “I knew the system would need to sound great and be robust, yet remain flexible enough to handle any changes or developments as the campaign progressed. With so many elements of the show connected to this rig, it could not be the weak link.”
Sanderson continues, “I’m very lucky that Ellie Goulding’s musical director, Joe Clegg, and the musicians in the band are all very open to embracing new technology. Together, we’ve designed a rig that allows us to explore both the creative and technical possibilities. I needed to select hardware that would not compromise the band’s workflow and give them the performance they needed while also providing the quality, reliability, and flexibility that I require to function efficiently on the road.
“To address these requirements, the RME and Ferrofish combination was a natural choice. The gear neatly handles all of our keyboard sounds, electronic drum sounds and playback, as well as autocue and timecode. In the end, we selected four RME MADIface XT units, the RME MADI Router, DirectOut EXBOX.BLDS MADI switchers (for auto-switching redundancy) and the Ferrofish A32.”
Sanderson provides specifics on the application of the Ferrofish A32. “The sound quality of this converter is fantastic and it couldn’t be easier to operate,” he reports. “We were one of the first people to get hold of it and we didn’t have a lot of time for extensive testing. As it turns out, it really was a case of unboxing it, bolting it into the rack, and turning it on. Straight out of the box, it worked exactly as you’d expect it to. The Ferrofish A32 is a very well thought out and intuitive piece of equipment. And while I’m certain we’re only using it for a fraction of its capabilities, for our requirements, it couldn’t be better.”
“With other artists I’ve worked for,” he adds. “I had a lot of success using RME products, so I knew the gear would work well. We needed equipment that was going to be robust, not compromise on sound or build quality, and give us the flexibility we required. The gear needed to be scalable up and down for different touring requirements.”
Sanderson also clarified why he and his team chose to go with the RME MADIface XT over the smaller RME MADIface USB, which was also considered. “Because this setup was never going to be a carry-on fly rig, compact size was less important. A key reason for choosing the XT is the unit’s visual display, which makes operation easier and speeds up problem diagnosis.
“Another important consideration is that I have the option to run them from the thunderbolt port via an adapter. This way, I can free up the USB architecture within the computer should it need to focus solely on MIDI.”
Sanderson reports that using MADI provides him extra flexibility and more sophisticated channel routing options. “We were quite ambitious with what we wanted to achieve regarding channel routing,” he notes, “and it was these products that enabled us to commit to a design without having to feel constricted by hardware specifications.”
He also points out the importance of having redundancy options in a live playback rig. The team uses two duplicate playback and keyboard rigs, ensuring that, if a fault were to occur during a show, the backup rigs would immediately take over, resulting in no audio dropouts. “
There are two sides to this system,” he says. “There’s a keyboard rig and a playback rig that also provides all of our electronic drum sounds and timecode, which gets sent out to FOH, as well as the Lighting and Video departments. Both sides to the rig have redundant back-up systems, but I also wanted the flexibility to be able to run the keyboard rig from the playback rig if necessary. The extra security provided by this system breeds confidence in the setup—for everyone involved.”
The rig is positioned offstage, though it is remotely controlled by the band on stage. The musicians are in total control of the show. There are five keyboards and two sample pads, plus nine drum triggers on the drum kit—all of which communicate directly with this system via MIDI. “I’m basically just monitoring the rig during the show, though I can make adjustments on the fly as required,” he says.
Friday, August 05, 2016
SSL AWS 948 Console Making Music At The New Anexe Studio In The UK
Owners Steve and Lindsey Trougton decided that the SSL AWS 948 could draw a wide crowd in addition to offering the desired functionality
The Anexe Studio, a new ground-up commercial facility in Exeter, UK, has invested in a Solid State Logic (SSL) AWS 948 console.
Steve and Lindsey Trougton’s long-held ambition to own and operate their own commercial recording studio has finally been realized with The Anexe. Returning from a spell in New York, where the pair originally went in 2011 to study sound recording after a dramatic change in careers, they saw opportunity in the region’s growing cultural credentials.
“It’s an up-and-coming city with a vibrant music and arts scene,” explains Lindsey. “There’s a lot of investment and growth, so we thought it would be an exciting place to build our studio, and our home.”
Both come from musical backgrounds. Steve is a trumpet and bass player as well as a vocalist, and has been in many bands from the age of 12. Lindsey is a vocalist who has trained in stage performance and spent a lot of time in recording studios over the years.
“There’s no direct competition for us here, which we thought was quite exciting,“ continues Lindsey. “We’re only a two-hour train journey from London and we have a local airport with regular flights to and from Europe.”
They decided that the SSL AWS 948 could draw a wide crowd. It uses a dual path channel design to fit 48 channels into a 24-channel frame, and offers three operating modes, selectable per channel. “To have 48 channels in such a small footprint is fantastic,” Steve says. “That was a massive selling point.
“It’s got 4-band EQ on every channel, selectable between E and G-Series,” he continues, “and SSL dynamics, and of course the master bus compressor from the G-Series console. I love that.”
The hybrid nature of the AWS is another aspect that the duo finds to suit their purposes well. The console’s focus button selects analog and DAW focus modes, allowing fast switching between an analogue and DAW operator focus, with reassigned meter, fader, select switch and V-pots. “Even when we’re tracking we’re on that Focus button doing the rough mix,” Steve explains. “Which is pretty much ready to go when the band has finished a take. That workflow is incredible.”
The Anexe Studio was built from scratch, starting with a 2.6-meter hole so the building could partially submerged into the landscape, giving it 4-meter-high ceilings in the live room without imposing too much on the local skyline.
UK studio design and installation specialists Studio Creations came up with the architectural, acoustic, and technical plans for the studio, and completed the construction, internal finishes, and technical installation.
Mark Russell, studio creations director, is complimentary about Steve and Lindsey’s vision for the studio: “The flow of the place, the vibe, everything is high quality. They have a fantastic eye for style and were very clear on their preferences. It’s not a ‘typical’ or ordinary studio. There’s a great mixture of materials used inside, including the Cedar paneling taken from the original building, the New York-inspired brick work, fabric panels. Lots of things.”
To help with the design process, the Studio Creations service includes 3D renderings of the designs plus an ‘auralization’ demo, which allows clients to assess different levels of sound proofing and isolation before committing to anything, and to satisfy any planning concerns.
Inside, the studio consists of a control room, a main live room with three isolation booths, and a recreational room with kitchen facilities. Even the bathroom sink is special, a one-off moulded concrete basin created from the body of one of Steve’s Telecasters.
The Trougtons’ have been putting the new facility through its paces with a stream of local acts. “We’re getting to know our space at the moment,” says Lindsey. “It’s been a learning experience for us and we’ve had some great feedback.
“Bands love the fact that they can record live. The local laptop studios have their place, but there’s no substitute for a dedicated live space and great equipment. Everybody knows the SSL name, though of course not all bands have experienced an SSL. They all love the look of the AWS 948, but they especially love the sound.”
Solid State Logic (SSL)
Posted by Keith Clark on 08/05 at 11:24 AM
Wednesday, August 03, 2016
The World Of Audio Splits
A live concert or event often serves a wider audience than those seated in the arena. In addition to the audio in the house and video at the side of the stage, it might also be simulcast to a separate location, streamed on the web, recorded for archival or other purposes, and/or broadcast via radio or television.
To accomplish this task, audio signals from microphones and instruments must be split beyond the traditional front of house and monitor positions and sent to other locations, be they other control rooms at the venue or inside production trucks.
Complicating matters more, these locations will probably be on different legs of the electrical system, potentially adding ground loops and hum to the signal.
With the more widespread adoption of digital consoles and networking in live sound applications, there’s much more potential to access the same signals and be tied together in ring or other topologies with the audio remaining in the digital domain. I was curious whether (and how) these newer technologies are being used.
My search began at the Monterey Jazz Festival, where the main stage shows are simulcast to the Jazz Theatre and other locations onsite as well as often shared via radio or webcast. McCune Audio (San Francisco) handles the audio and video for the festival. I then talked with folks at several major sound companies about how they handle the issues of signal splitting, grounding, and interfacing consoles.
Main Stage Simulcast
The 6,500-seat arena at the Monterey Jazz Festival, where the main acts perform, is the focal point of the simulcast and broadcast activities. DiGiCo SD10 consoles are located at FOH and monitors, with their D-Racks at the side of the stage near the monitor station. Four video cameras are positioned in the house for full-stage shots as well as on either side of the stage for close-ups.
“Split world” adjacent to the monitor position at the Jazz Theater at Monterey, the focal point of simulcast and broadcast activities. (Photo by Eva Bagno)
The video and simulcast control center is set up behind and underneath the main stage. All camera shots are called from this location, audio is embedded with the video, simulcast feeds are processed and distributed, and in 2014, the show was also webcast. Archiving is done from a production truck outside the arena, with an Avid VENUE Profile console building its own mix.
This mix is also sent to radio, as it has been every year with the exception of the 2015 edition of the festival. The raw feeds from the onstage cameras are provided to screens at both FOH and the archiving truck so that the engineers have visual information about what’s happening on stage, even during changeovers between acts.
Splitting & Routing
To accommodate the different mix requirements for FOH, monitors, archiving, and simulcast, stage signals are split three ways.
Though all mixing consoles are digital, with A/D converters on stage, the split itself is analog via a custom Ramtech STGBX-54 three-way splitter. Each of the 54 channels can be input directly or via four 12-channel and one 6-channel Ramtech CPC onstage sub-snakes.
The SD10 console at FOH is directly connected, and the monitor and archiving consoles receive transformer-isolated feeds. “Any mic that’s in the system, even if it’s only for a record input, has to connect to FOH for phantom power,” adds Nick Malgieri, FOH mixer.
From the splitter, the stage signals go from each of the three multi-pin outputs to the A/D converter boxes. Both FOH and monitors use two 32-input DiGiCo D-Racks, with an optical fiber loop to transmit the signals to the consoles. To feed the Profile console for archival recording, the third split goes to a VENUE Stage Rack using MADI digital protocol. The video control area receives two separate stereo mixes, from FOH and the archival truck, and uses one or the other to embed with the live-edited video signal – which goes from there to simulcast, recording, and webcast.
On the decision to use an analog split rather than digital networking to share audio signals, Malgieri explains, “There are ways we could network it all together, but for speed and efficiency we keep it an analog split. That way, no one is tied to anyone else. Sharing preamps together would make us interdependent, which wouldn’t be conducive to a festival-style event with fast changes and guest engineers.”
Recording & Archiving
Ron Davis has been mixing and producing the Monterey Jazz archival recordings for many years. The mix is independent of FOH, starting with the raw signal “straight off the mic” plus the audience mics above the stage and in the house for crowd response. Having his own multi-channel feed from the stage allows him to “fine-tune the mix for recording purposes,” he notes, since his environment is more conducive to critical listening.
Ron Davis, mixer and producer of Monterey Jazz archival recordings, at his Avid VENUE Profile console. (Photo by Eva Bagno)
The VENUE Profile console interfaces with Pro Tools, and Rob Macky monitors the recording along with other technical details. Other members of the team include an onstage liason, who is in touch continually with a comms person in the truck (also connected with FOH, video, and other positions throughout the venue), and another who archives the recordings to digital media as soon as they’re finished.
Davis says that 48 channels are usually more than enough for the acts plus the audience mics, though at times he needs to drop a couple of inputs. In those cases, he may choose one of a stereo pair of mics or just use the DI from the bass rather than adding the mic on the cabinet. His mix is patched to the video area as a potential simulcast feed, and is fed to any radio broadcast trucks airing the show. The archival truck also receives the FOH mix for redundancy.
Monterey Jazz Simulcast
Because simulcast is an important feature at the festival, a control area is designated for video and tasked with live video for the side-stage screens, creating the simulcast feed and monitoring the venues where it plays, video archiving and performance MP4 recordings for the artists, and at times, webcasting.
The crew includes the camera operators, directed by Jesse Block from the control room, a person controlling the video and audio embedding, another on recording, and a “grounds technician.”
A portion of the simulcast control center underneath the main stage at Monterey Jazz. (Photo by Eva Bagno)
Simulcast receives mixes from FOH and the record truck, plus an ambient mic submix. Malgieri states that “Depending on who’s ready first, video makes a judgment depending on what’s coming down the pipe on which mix they’re going to go with. This could change for each act.” Also, having both mixes available provides a backup in case of trouble, giving the same content but different mixes.
The simulcast feed goes to the Jazz Theatre, where patrons who have purchased ground passes for the other stages can experience what’s happening on the arena stage. The signal is sent about 750 feet via fiber to the theatre, and then decoded into L/R audio to full-range loudspeakers and subwoofers, with video projected on a large screen. The Premier (VIP) Lounge is a smaller venue, closer to the arena, and it also receives the simulcast via HD/SDI.
On The Road
Beyond Monterey, I checked in with several other touring companies to learn how they handle signal splitting for live events, especially when multiple splits are required for broadcast, recording, or similar applications.
In most cases, an “old-school” analog splitter with transformer-isolated outputs is the rule on the road (at least among those I spoke with), rather than sharing a common digital signal among the various applications.
Dave Skaff, senior tour support for Clair Global (Lititz, PA), says that “There seems to be two distinct camps between the live world of traveling music and broadcast. The live mixers are very ingrained with having their own head amp control. To give them that control, a digital split is kind of ruled out.”
A downside is that each console position needs its own stage racks with an analog split. Continuing, he adds, “In the broadcast world, the idea of using one set of head amps, and having several people follow with digital trim or some kind of gain tracking is a fairly accepted way of doing things – their comfort level is higher.”
On a recent U2 tour, Skaff notes, “We did entertain the idea of having digital splits, with certain people having control of stage racks and others using digital trim for levels.” During the planning, he adds that there were incidents where, if the digital loop went down, “you lost a lot of control.” The show was especially complex, with six different consoles that would be on the loop – FOH plus a backup, and three separate monitor setups with a backup.
Skaff notes that some of the tour staff’s fear of relying on the newer technology came from second-hand conversations they’ve “heard from others,” plus Clair’s own observations of small glitches that persuaded them to stay with the tried-and-true methods.
The company went back to a custom-designed 6-way analog splitter for the U2 shows so that each console would have control over levels, with proper loading for six mic preamps per channel and transformers that accept a wide range of signal levels without saturation. There were also conventional splitters that facilitate 3- or 4-way passive splits.
For a show that also needs to accommodate broadcast trucks, the engineer might ask for an isolated analog split or a digital split sent to them as AES3, or possibly a MADI split off the stage racks; Skaff has had all of the above requested recently. Many tours will provide an open analog split, available for a recording truck or other production application.
Smaller Or Larger Flexibility
Dave Rat of Rat Sound Systems (Oxnard, CA) discussed with me a “baseline method” of signal splitting, using a custom-built XLR panel with two 56-pair Whirlwind W4 MASS connector outputs. For smaller shows, the choice is usually a single panel for FOH and monitors, while for larger shows or ones that require separate recording or broadcast feeds, the approach is multiple panels with a single input and a pair of ISO outputs – with the direct signal going to FOH.
A portion of the isolated split recording approach with the Red Hot Chili Peppers in Europe earlier this year.
Rat Sound has also designed multi-connector panels that are fed from stage boxes, and by changing the tails, the switch between opening and headlining acts can be accomplished more quickly and reliably. Occasionally there are bands where both the FOH and monitor engineer are working with the same mixing console, and each will use a common digital split; any additional feeds for a production truck are likely to come from the analog splitter.
Rat observes that recording seems to fall into two categories.
The first is an isolated split recording where the signals go to a recording truck, or in the case of the Red Hot Chili Peppers, to a console located in a remote room in the venue that is fed from a separate A/D rack stage-side and mixed there. This mix might go to broadcast or another application.
The second is that the FOH or monitor console sends a recording feed, taking the mic outputs in their raw form to a multitrack recorder.
Greg Snyder of Thunder Audio (Livonia, MI) also confirms that even though many of the latest mixing consoles can share a stage rack and digitally split the signal, his team often opts not to share mic preamps and to use an analog split. He finds that off-the-shelf splitters can be very reliable, and that “with today’s digital consoles, we find that passive splitters are very easy to use as go-to packages.”
Snyder adds that quite often the FOH engineer will create a mix to be embedded with video, which is transported from the console to the video truck via an analog snake or fiber interface. He notes that mixing for both live and video “requires that the engineer be very conscious of the mix they’re providing so that it will be usable for broadcast.”
Hall Of Fame & More
When I caught up with him recently, Mark Dittmar, the live broadcast events engineer at Firehouse Productions (Red Hook, NY), had just returned from the Rock & Roll Hall of Fame show, which he’s worked for several years running. This year’s event combined a live show for about an audience of about 15,000, plus broadcast, at Brooklyn’s Barclay Center arena.
In addition to Firehouse’s live audio setup that included infrastructure, splits and comms, All Mobile Video provided the television truck and a Music Mix Mobile truck did the audio mixing. Firehouse handles overall coordination of the show, and then, Ditmar notes, “informs the others how we’re handing things off to them.”
Both 3-way and 6-way splitters were deployed to route audio signals to the various stage racks, and then to mixing consoles in the venue and out to the trucks. “We always split everything analog; we don’t do any digital sharing,” he explains. “That has a major negative impact on the speed and workflow that we’re doing. We keep everything analog in the split world, and then it goes digital from that point on out. With how fast changes come at us in this type of show, it’s proven to be impractical for any type of preamp sharing.”
He also points out that the FOH music and production desks, monitor desk, music mix desk, and broadcast desk are usually different makes/models, and there’s typically only one song during rehearsal to set levels and EQ, along with a quick camera check, and then it’s on to the next act.
Firehouse utilizes modified Whirlwind splitters, with Dittmar noting that “one of the cornerstones of our company is having absolutely zero split issues.” It’s not uncommon to see a 192-input show split six ways, so the company uses a very specific grounding scheme and is “militant” about sticking to it.
Part of the splitter’s design is focused on enforcing proper grounding, and the tech crew also follows a rigid power distribution scheme that also reinforces best practices in grounding. Dittmar concluded our conversation by stressing the basics: “Splitting and grounding is something where you can be 99.9 percent correct, and the 0.1 percent that you’re wrong about brings everything down.”
Grounded In Analog?
While digital networking has matured greatly over the past several years and can effectively distribute audio signals to multiple sources reliably, there are still some areas within the audio chain where old-school analog devices remain a standard. Signal splitting and isolation seems to be one of those areas.
In part, this is a practical decision driven by the nature of shows being set up in different venues every night while accommodating the rapid changes between acts and the desire of engineers to have full control over the inputs into their consoles. There also seems to be some resistance to surrendering that control, based on prior experiences with earlier networking technology and anecdotes from fellow engineers.
The bottom line is that analog splitting is a proven solution for sharing live audio. It will take more time, positive experiences and perhaps technical development before digital splitting becomes more commonplace in live sound.
Gary Parks is a writer who has worked in pro audio for more than 25 years, holding marketing and management positions with several leading manufacturers.
Tuesday, August 02, 2016
A-Designs Audio Unveils Mix Factory Summing Unit For Enhanced Workflows
Accommodates up to 16 audio channels that come into the device on two D-sub inputs and sum to stereo XLR outputs
A-Designs Audio has introduced Mix Factory, a new concept and approach to “out-of-the-box” summing for engineers and musicians looking to enhance their current sound and workflow.
“Our new Mix Factory isn’t just any old summing unit,” says A-Designs Audio’s Peter Montessi. “It delivers analog warmth with the depth and imaging needed to make your mixes truly stand out from the crowd.”
Based on a concept developed by producer/engineer/mixer Tony Shepperd and brought to life by designer Paul Wolff, Mix Factory accommodates up to 16 audio channels, which come into the device on two D-sub inputs and sum to stereo XLR outputs.
All 16 channels have a continuous FDR (gain) knob, pan pot with center detent, and cut (mute) switch that acts as a signal indicator with an audio sensitive LED, which glows when signal is passing into the channel and intensifies when the signal is stronger. That same cut button illuminates red when used as a mute, and green when signal is passing through the channel.
There are two eight-channel groups on the Mix Factory: 1-8 and 9-16. Each group has an insert for a compressor or EQ, and there is also a master insert for all 16 channels, along with three mute buttons for each Insert.
Mix Factory has a push-button option to go from clean—the standard setup bypassing the transformers—to tonal using the custom-made output transformers manufactured by Cinemag. The difference between using the transformers and not provides users an option for analog tone and/or color.
In situations where more than 16 channels are necessary, Mix Factory is linkable, providing 64 or more channels.
“Like many other present day engineers, I used to think that everything could be done ‘in the box’,” recalls Shepperd. “However, I finally came to realize that analog and digital could, in fact, co-exist very well together, and the hybrid of the two was the special combination that took my mixes to a whole new level. It’s what ultimately prompted many other recording engineers to call me for tips and practically demand to know how I made my recordings sound so great. After several years of R&D on this product, I can say that nothing will make a mix pop quite like A-Designs’ new Mix Factory.”
An external, switchable power supply allows the unit to be used for both 120-volt domestic (U.S.) studio environments as well as 230-volt export markets. Including the PSU, the 2U device weighs 17 pounds.
Immediately available, Mix Factory carries a street price of $2,750 (USD).
Thursday, July 28, 2016
Analog & Digital Consoles
One day on a freelance gig I walked into the room to discover that the A/V company provided me with an older analog console with two racks of outboard gear. While setting up front of house and patching in all of the effects and processing, I found myself wishing for a digital console.
The very next freelance gig I was presented with a brand-new digital console and no extra gear at front of house. As I waded through menus trying to set up a console that I ‘d never used before and struggled to read the manual I’d just downloaded on my phone, I found myself wishing for an analog console and some simple outboard processing.
The death of analog consoles and processing hasn’t happened yet, nor will they go away in the foreseeable future. Many sound companies and installations still utilize analog components, and there are “old sound guys” like me who still like to grab knobs instead of sort through menus.
For shows with just a few inputs, analog is usually the most cost-effective choice, and for some larger shows they’re a proven item that (often) sound great while offering all of the necessary features needed.
If you work in only one venue, you learn the console and system, and have adapted a workflow for that gear. If you freelance a lot like me, there’s the need to adapt to whatever console and equipment is provided. Here are some of the approaches I’ve developed over the years.
I have slightly different setup routines depending if it’s a digital console or an analog model with outboard processing. The first thing I do, if at all possible ahead of time, is ask what gear is being provided so I can download the manuals and quick start guides if I don’t already have a copy. There are folders on my laptop filled with manuals for consoles, outboard processors and recording units, so answers can be found quickly at a show even if I don’t have internet access.
If I find out about the gear in advance then I print out a quick start guide and/or pertinent pages of a manual (like how to change aux sends from pre to post) for quick reference at the gig. It’s also a good idea read the manuals on unfamiliar gear before the show.
With analog consoles at FOH, I start by setting up the outboard racks close to the console so the patch cables can reach. Depending on how the equipment is packaged I may place the console on top of the outboard racks to save space or put the outboard racks off to the side at a 90-degree angle.
Next up is figuring out what’s needed with respect to inputs, outputs and processing. Digital consoles usually have processing available for every input and output, but analog requires a bit of forethought, including strategies and “workarounds” when processing channels and devices are limited.
Digital consoles, particularly more recent models, carry an impressive amount of processing onboard.
Patching It Up
Once the game plan is ready, I kick things off by patching the outputs for the main loudspeakers as well as any delay and fills. This involves running short patch cables between the console outputs and the outboard EQs in the rack, then patching the amplifiers (or powered loudspeakers) from the outboard EQ units.
There may be the need to patch in a delay unit as well if running remote loudspeakers. Once the loudspeakers are patched, it’s time to can address any output feeds that are needed, such as an audio send to video world or a feed to lobby loudspeakers.
Next up is patching in effects and channel processing like compressors and gates. Most analog systems that I run across have between two and eight channels of compression available, so it’s a good idea to plan ahead with a limited number.
With analog consoles, outboard processing is usually “inserted” into a channel via an insert jack. Some consoles have two send and return connectors that are usually 1/4-inch TRS (balanced) for inserting external gear, but more common is a single 1/4-inch TRS insert jack that provides an unbalanced send and return on a single plug.
A special “insert cable” that consists of a 1/4-inch TRS plug on one end that breaks out into two 1/4-inch TS plugs at the processing end is used to interface the outboard gear.
The cable is wired so the Tip of the TRS is the send to one of the breakout legs, the Ring is the return on the other breakout leg, and the Sleeve is wired to the sleeves on both breakout legs. I always carry some insert cables to shows as they seem to be the most forgotten item on the pack list.
Now it’s time to plug in the inputs and label the console. I also take the time to label the aux sends as well as the outboard gear so I know where they’re patched in.
A Bit Easier, But…
Digital consoles are a bit more simple to set up as most offer the necessary processing onboard. Before doing anything with a digital desk, I like to start from scratch, wiping it back to the factory settings. Some models have a default scene that can be recalled, some require a few clicks in the menu, and others may require a boot-up while holding down a few buttons to reset back to the default.
An insert cable with 1/4-inch TRS plug and two 1/4-inch TS plugs.
My reasons for starting with a clean slate are simple. I don’t know what the last user did and don’t want to get “bit” during my show trying to make an adjustment only to find out that the last user changed a setting that was not readily apparent, like switching all of the auxes to post fader.
Once the console is reset, I still start at the outputs and make sure there’s an EQ in-line with the loudspeakers. Many models have an EQ assigned to each output. but a few require assigning any needed processing to a specific output from a limited number of items onboard.
After outputs comes inputs, and I assign each channel the processing it requires. Again, some boards have a limited number of processing and effects, and these must be chosen and patched before they can be used. If the console has scribble strips, label each input and output as you go. I also like to label the console with tape, especially for items that don’t have scribble strips like user-assigned buttons. At this point it’s also a good idea to save the settings on the console in a scene.
If I’m using a remote app for mixing on a tablet, now is the time to make sure it’s working and then walk around the venue to see if it stays connected around the room.
On most corporate gigs and small festival band gigs, the supplied analog console usually has between 24 and 32 inputs and four or eight subgroups. Larger shows may get a larger frame board with 40-plus inputs and VCAs. Since VCAs aren’t available on many shows, I use the subgroups to make my job a little easier by grouping like inputs together.
With four groups available on corporates, I normally place the podium mic into its own group, any presenter wireless into a group, table microphones for panel discussion into a group and Q+A audience mics in another. With four subgroups available on band gigs I split up the groups into drums, guitars, keyboards with horns (if any) and vocals.
It can help to add a bit of compression to a podium and lavalier mics used by presenters. With music it’s likely to add some compression to kick drum, bass guitar, lead guitar and vocals.
I work with a lot of headline singers who have some background vocalists behind them. One trick that comes in handy when there’s not a lot of outboard compression channels is to run all of the background vocals into a subgroup and compress the group as a whole, but insert a compressor into the lead singer’s channel and then just run them straight to the L+R masters.
That way there’s tailoring available for the lead vocalist and the ability to add compression to the background singers, and without taking up an entire subgroup for one vocalist.
On corporate gigs it’s not uncommon for me to get stuck with a very small analog console with limited channel EQ but there’s a high-profile presenter who needs some drastic EQ help to sound good in the room. If the console has a channel insert jack, I insert an outboard graphic EQ or parametric EQ into the channel.
I carry a stereo 15-band graphic just in case there’s not an extra one provided onsite. This EQ has often come in handy for the mains when the A/V company thinks that a small console with built-in five- to seven-band graphic is all you need and they don’t provide an outboard EQ.
Other outboard gear regularly carried to freelance gigs include a stereo leveler, stereo compressor/gate and multi-channel feedback suppression processor. While some audio folks laugh when they see my feedback unit, they soon realize that it does a great job taming wireless lavalier mics because each of the 24 filters can have a bandwidth as narrow as 1/80 of an octave. It works great when inserted on the lavalier subgroup.
Another limitation on some smaller analog consoles is a lack of buses. More than a few times I’ve been mixing monitors from a smaller front of house board and have run out of aux sends. If the console has a matrix section, it can be used to set up side fill mixes, and in a pinch, some individual performer mixes.
Even the smallest digital consoles usually include comps and gates on every channel but many of them lack VCAs (also called DCAs) or subgroups. An easy way to get subgroup control on a digital board that does not offer groups is to use a post fade aux bus. Simply assign every input channel that you want in the group to the same post fade aux send. Make sure to un-assign those channels from routing to the main L+R outputs. Now assign the output of that aux send to the L+R mains and it acts as a subgroup.
While the aux master may be on a different layer, many consoles have a user-configurable layer that can be tailored to specific needs. For example, I place my “money channels” like lead vocalist, lead guitar, podium, presenter wireless and others on the user layer, along with any effects masters and subgroups or VCA/DCAs. This way, mixing can be done mostly on a single layer.
One of the few drawbacks with digital desks, at least for me, comes when using a digital audio network instead of an analog snake. The problem is that an intercom channel or lighting DMX can’t be run on the same cable as can be done if the analog snake has extra channels. Sure, the solution is running a few single cables from FOH to backstage, but finding hundreds of feet of extra XLR cable onsite can be an exercise in futility and carrying around hundreds of feet of “extra” cable as a freelancer is not an option.
Catapult, a recently released 4-channel Cat snake from Radial Engineering.
My solution is to deploy a snake system that provides four analog audio runs down a single shielded Cat cable. A small reel with 100 meters of cable makes for a compact package I can keep in my truck. Radial Engineering, Whirlwind and others offer nice 4-channel snake boxes for Cat cable. In addition to comms and DMX, they work great for analog mics, line level returns and even AES signals.
Speaking of networks, Dante has become the de facto standard of audio networks, so I carry a few small gigabit switches to shows to help route signals. I also just added a small wi-fi router to my bag. It comes in handy for interfacing an iPad for remote console control and might also come in handy with the new version (3.10) of Dante Controller able to connect to networks over wi-fi.
Finally, for almost every gig, I bring a small utility mixer just in case the console has a problem or a submixer is needed (or would come in handy). In the past this was an 8-channel analog unit but recently has switched to a very compact digital mixer. With 16 inputs that have 4-band EQ and onboard multitrack recording, I sometimes just replace the console provided by the A/V company with my own mixer and a small rack.
The key as a freelance audio technician is to be able to adapt to the equipment that is provided no matter if analog or digital, and to be prepared outside of that to make the show happen, and as well as possible. Making the client happy insures I get another call.
Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas.
Tuesday, July 26, 2016
Precision Sound Studios Steps Up To Solid State Logic
Malvicino Design Group outfits Manhattan recording facility with 48-channel SSL Duality δelta Pro Station SuperAnalogue console.
Precision Sound Studios owner and engineer/producer Alex Sterling has created a technical and creative oasis on the Upper West Side of Manhattan, close to the City’s American Museum of Natural History, Central Park, and many cosmopolitan shops and restaurants nearby.
As part of a recent re-fit and technical upgrade package, Sterling has installed a 48-channel Solid State Logic Duality δelta Pro Station SuperAnalogue console in the Precision Sound Control Room A.
The Studio upgrade, specified by Sterling and implemented by the Malvicino Design Group, also included a comprehensive new wiring scheme, a large video screen for Film/TV Post work, and overall layout adjustments and refurbishment.
“I have always wanted to create a working space for music production that has the comfort of a person’s home or living room but with the technical and professional capabilities of a larger commercial facility,“ explains Sterling.
One of the most striking features of the studio is the live room space, which also happens to be a library of around 3000 books. “Believe it or not,” says Sterling, “They have an acoustic value as well as an aesthetic value.”
The live room can host around 15 musicians, which means that Sterling is as in demand for band recording, film, and television work as he is for Electronic, Pop, and Hip Hop mixing and production. For Sterling, the new Duality δelta console is a creative tool that meets his own high standards, yet also puts his studio onto a more high-end commercial footing with outside producers.
“During my console search I carefully researched and demoed several other modern consoles, many of which did have some substantial sonic attributes, however the Duality has the most developed functionality for a modern workflow and its sonics are nothing less than spectacular.
“The integration with the DAW was very important to me, as was the high channel count - and having a full complement of processing available on every channel… I could be spending twice as much to get full filters, dynamics, and EQ on every channel with another console, and I still wouldn’t be getting any of the DAW control functionality that Duality offers.”
Precision Sound has now been up and running with the new console for several months. The very first session on the new console was a TV scoring session for composer Michael Bacon. “That was a good first test,” says Sterling. “Everything was flawless, everything sounded great…
“I’ve used the console’s channel preamps for most of the tracking that I’ve done through the desk… I was not expecting to like the preamps as much as I do. For tracking, the SSL pre-amp is as transparent as any of the esteemed, clean boutique pre-amps, and it’s extremely low noise, which some other pre-amps just can’t claim.”
Sterling is also complimentary about the SuperAnalogue bus architecture of the Duality.
“One of the things I’ve been experimenting with is using the console’s mix bus to give me volume and level for a final mix print, but without having to use peak limiting. By driving the console mix bus with a lot of level, I am able to get a much more aggressive full and forward sound - without needing to lose or cut off transients with a dynamics processor for volume.
“I’ve been shocked how rich and full I can make things sound by essentially ignoring the VU meters and letting them pin completely into the red… just completely brutalizing the capture chain. “...The desk can really take it. You can clip the channels a bit, but the mix bus itself is pretty much unclippable. At least, I haven’t managed to do any damage with it yet…”
For Sterling, the last few months have proven that the Duality delivers superb sonics, an integrated DAW workflow, and a creative approach to production.
“To my ear, signal processing is generally superior in the analogue domain,” he says, “But some of the creative things that people are doing now really only exist within the DAW environment. To not become disconnected from the DAW while working on the console was very important to me because I’m working on modern productions that have modern production requirements… This console really has set the professional standard for this decade.”
Solid State Logic
Malvicino Design Group
Thursday, July 21, 2016
Sandlane Recording Facilities Steps Up To Rupert Neve Designs
Maarten de Peijper's recording studio in the Netherlands upgrades to 32-channel 5088 console with Shelford 5052s.
Located in the southern part of the Netherlands, Sandlane Recording Facilities was established in 2009 by Maarten de Peijper as a recording studio designed to feel “like a home away from home” with two control rooms, two live rooms, and a living space.
Equipment-wise, Sandlane is outfitted with an array of equipment from Tube-Tech, Retro, UREI, Drawmer, API, Chandler, Maselec, Empirical Labs, and vintage Neve modules – but the studio’s heavy lifting is now handled by a 32-channel Rupert Neve Designs 5088 console, loaded with Shelford 5052s.
“Next to a collection of high-end gear and a great-sounding live room, we pride ourselves on having a pleasant atmosphere. We believe that when an artist feels no constraints from the environment, they can fully focus on getting that perfect take,” says de Peijper.
“After being in business for a few years, we found that our previous console became the bottleneck in everything we did. After some looking around for a serious upgrade, we fell in love with the sound and ease of use of the 5088…we compared the 5052s to a couple of Neve 1064 units, which were our most prestigious pre-amps at the time. The sound and musicality were so close we decided to sell the 1064s and fully load the console with 5052s.
“The 5088 is very much the heart of the studio…it has completely turned our workflow inside out. It has an insane amount of headroom for summing. Even with complex, layered and heavily compressed music maxing out the stereo bus, it retains its openness. Also, the EQ’s of the 5052’s are amazing; they help shape the sound without compromising the integrity of the source. Everything can be shaped and still sounds completely natural.”
Sandlane’s 5088 is configured with 24 mono and 8 stereo channels. When recording, the mono channels feed into the A/D converters and the stereo channels are used for listening back to submixes from Pro Tools.
“This split setup is our standard way of recording, but because every input and output of the console is hooked up to our patchbay, we are able to switch the console to an in-line setup within a matter of seconds. When it’s time to mix, we simply flick the mic/line switches on the 5052s. That way we have the 24 mono channels receiving input from Pro Tools, ready to be used in combination with our outboard gear – and of course that beautiful analog summing.”
Sandlane recently finished 5 months of recording a Dutch symphonic metal band called Epica under the guidance of producer Joost van den Broek, with some songs consisting of over 700 tracks – strings, brass, percussion, and choir layered over a full metal production. What comes after that?
“We have been steadily growing over the past years, both as a studio and as engineers/producers. Our relatively small team is in it for the long run and we’re all dedicated to following this positive course. Hopefully we can keep making bigger productions and keep having a lot of fun in the process.”
Rupert Neve Designs
Sandlane Recording Facilities
Posted by House Editor on 07/21 at 06:46 AM
Wednesday, July 20, 2016
Germany’s Parkhaus Studio Returns To Analog With SSL
Engineer Albert Gabriel adds AWS 948 hybrid console with SSL SuperAnalogue architecture and DAW control to studio in Köln.
Originally a low-key haven for songwriters and producers, Parkhaus Studio has been a creative force in Köln, Germany for 10 years, becoming more commercial in recent times with acoustic and technology upgrades.
Albert Gabriel, recording and mix engineer/chief tech at Parkhaus, moved in and extended the range of services on offer, bringing a Solid State Logic AWS 948 to the facility as part of his own move to a more analog outlook.
He sees this as major step forward: “I had been working ‘in-the-box’ for the ten years before I bought the AWS,” he says. “One day, a friend asked me to work on something… Something that couldn’t work without analog gear… I realized that what I did in back the nineties on an analog console was way more natural than everything I had done after that.”
Gabriel went back to his roots (even acquiring a two-inch multitrack tape machine) and eventually found the SSL AWS 948 - a hybrid console with SSL SuperAnalogue architecture and DAW control. “Now Parkhaus is growing particularly fast,” he continues. “With an SSL in the studio the old customers are coming back, and telling more customers to come too.”
Gabriel describes Parkhaus as a ‘musician-friendly’ studio, with plenty of space, and a relaxing environment. Facilities include a large central live room connected to two control rooms, each of which have their own vocal booths that double as production rooms, stocked with synths, guitars, and more. The main control room has the AWS 948 and an assortment of original analog gear.
“I know the SSL computer and the automation from the older consoles, so I found it really comfortable to switch to the AWS. Sound-wise I think it’s pretty close the K Series - and I never had a better feeling in a studio than with the K Series.”
The combination of superb SuperAnalogue architecture plus the AWS 948’s unique dual path channel strip design and its trio of operating modes means that Gabriel is as happy tracking and mixing with the SSL as he is mastering from stereo stems. “The bus routing options, the Stereo Mix mode, and the opportunity to insert the dynamics wherever you want them are all features that are great for mastering,” he says. “If there was an eight-channel version of the AWS, it would be called a mastering console.”
The AWS DAW control is also important to Gabriel. “I use it 24/7,” he says. “The plugin control is awesome. I often turn the DAW screen off when I’m mixing because I get access to all the plugins using the AWS screen. That’s better than the DAW control because you can concentrate on the center of the desk and the center of the control room. That’s always been an important concept for SSL consoles.”
Parkhaus continues to grow and continues to nurture the creative collective that is its expanding client-base. The SSL AWS 948 is an important part of that success story, along with Gabriel’s mission to keep sound at the top of the priority list - along with a flexible, welcoming space, expertise on-tap, and the best tools for the job.
Solid State Logic
Tuesday, July 19, 2016
Signal Processing Fundamentals: Passive & Active Crossovers
In space, no one can hear you scream ... because there is no air or other medium for sound to travel.
Sound needs a medium; an intervening substance through which it can travel from point to point; it must be carried on something. That something can be solid, liquid or gas. They can hear you scream underwater ... briefly.
Water is a medium. Air is a medium. Nightclub walls are a medium. Sound travels in air by rapidly changing the air pressure relative to its normal value (atmospheric pressure). Sound is a disturbance in the surrounding medium.
A vibration that spreads out from the source, creating a series of expanding shells of high pressure and low pressure ... high pressure ... low pressure ... high pressure ... low pressure.
Moving ever outward these cycles of alternating pressure zones travel until finally dissipating, or reflecting off surfaces (nightclub walls), or passing through boundaries, or getting absorbed—usually a combination of all three.
Left unobstructed, sound travels outward, but not forever. The air (or other medium) robs some of the sound’s power as it passes. The price of passage: the medium absorbs its energy.
This power loss is experienced as a reduction in how loud it is (the term loudness is used to describe how loud it is from moment to moment) as the signal travels away from its source.
The loudness of the signal is reduced by one-fourth for each doubling of distance from the source. This means that it is 6 dB less loud as you double your distance from it. [This is known as the inverse square law since the decrease is inversely proportional to the square of the distance traveled; for example, 2 times the distance equals a 1/4 decrease in loudness, and so on.]
How do we create sound, and how do we capture sound? We do this using opposite sides of the same electromagnetic coin.
Electricity and magnetism are kinfolk: If you pass a coil of wire through a magnetic field, electricity is generated within the coil. Turn the coin over and flip it again: If you pass electricity through a coil of wire, a magnetic field is generated. Move the magnet, get a voltage; apply a voltage, create a magnet ... this is the essence of all electromechanical objects.
Microphones and loudspeakers are electromechanical objects. At their hearts there is a coil of wire (the voice coil) and a magnet (the magnet). Speaking causes sound vibrations to travel outward from your mouth.
Speaking into a moving-coil (aka dynamic) microphone causes the voice coil to move within a magnetic field. This causes a voltage to be developed and a current to flow proportional to the sound—sound has been captured.
At the other end of the chain, a voltage is applied to the loudspeaker voice coil causing a current to flow which produces a magnetic field that makes the cone move proportional to the audio signal applied—sound has been created.
The microphone translates sound into an electrical signal, and the loudspeaker translates an electrical signal into sound. One capturing, the other creating. Everything in-between is just details.
And in case you’re wondering: yes; turned around, a microphone can be a loudspeaker (that makes teeny tiny sounds), and a loudspeaker can be a microphone (if you SHOUT REALLY LOUD).
Crossovers: Simple Division
Loudspeaker crossovers are a necessary evil. A different universe, a different set of physics and maybe we could have what we want: one loudspeaker that does it all.
One loudspeaker that reproduces all audio frequencies equally well, with no distortion, at loudness levels adequate for whatever venue we play.
Well, we live here, and our system of physics does not allow such extravagance. The hard truth is, no one loudspeaker can do it all.
We need at least two—more if we can afford them. Woofers and tweeters. A big woofer for the lows and a little tweeter for the highs. This is known as a 2-way system. (Check the accompanying diagrams below for the following discussions.)
But with two speakers, the correct frequencies must be routed (or crossed over) to each loudspeaker.
At the simplest level a crossover is a passive network. A passive network is one not needing a power supply to operate—if it has a line cord, or runs off batteries, then it is not a passive circuit.
The simplest passive crossover network consists of only two components: a capacitor connecting to the high frequency driver and an inductor (aka a coil) connecting to the low frequency driver.
A capacitor is an electronic component that passes high frequencies (the passband) and blocks low frequencies (the stopband); an inductor does just the opposite: it passes low frequencies and blocks high frequencies.
Above, passive 2-way crossover, and below, passive 3-way crossover.
But as the frequency changes, neither component reacts suddenly. They do it gradually; they slowly start to pass (or stop passing) their respective frequencies. The rate at which this occurs is called the crossover slope.
It is measured in dB per octave, or shortened to dB/octave. The slope increases or decreases so many dB/octave. At the simplest level, each component gives you a 6 dB/octave slope (a physical fact of our universe).
Again, at the simplest level, adding more components increases the slope in 6 dB increments, creating slopes of 12 dB/oct, 18 dB/oct, 24 dB/oct, and so on.
The number of components, or 6 dB slope increments, is called the crossover order. Therefore, a 4th-order crossover has (at least) four components, and produces steep slopes of 24 dB/octave.
The steeper the better for most drivers, since speakers only perform well for a certain band of frequencies; beyond that they misbehave, sometimes badly. Steep slopes prevent these frequencies from getting to the driver.
You can combine capacitors and inductors to create a third path that eliminates the highest highs and the lowest lows, and forms a mid-frequency crossover section. This is naturally called a 3-way system. (See diagram)
The “mid” section forms a bandpass filter, since it only passes a specific frequency band. Note from the diagram that the high frequency passband and low frequency passband terms are often shortened to just high-pass and low-pass.
A 3-way system allows optimizing each driver for a narrower band of frequencies, producing a better overall sound. So why not just use passive boxes?
The single biggest problem is that one passive cabinet (or a pair) won’t play loud enough and clean enough for large spaces. If the sound system is for your bedroom or garage, passive systems would work just fine—maybe even better. But it isn’t.
Once you try to fill a relatively large space with equally loud sound you start to understand the problems. And it doesn’t take stadiums, just normal size clubs. It is really difficult to produce the required loudness with passive boxes.
Life would be a lot easier if you could just jack everyone into their own cans amp—like a bunch of HC 4 or HC 6 Headphone Amps scattered throughout the audience. Let them do the work; then everyone could hear equally well, and choose their own listening level.
But life is hard, and headphone amps must be restricted to practice and recording.
Monitor loudspeakers, on the other hand, most likely have passive crossovers. Again, it’s a matter of distance and loudness. Monitors are usually close and not overly loud—too loud and they will feed back into your microphone or be heard along with the main mix: not good.
Monitor loudspeakers are similar to hi-fi loudspeakers, where passive designs dominate ... because of the relatively small listening areas. It is quite easy to fill small listening rooms with pristine sounds even at ear-splitting levels.
But move those same speakers into your local club and they will sound thin, dull and lifeless. Not only will they not play loud enough, but they may need the sonic benefits of sound bouncing off close walls to reinforce and fill the direct sound. In large venues, these walls are way too far away to benefit anyone.
So why not use a bunch of passive boxes? You can, and some people do. However, for reasons to follow, it only works for a couple of cabinets. Even so, you won’t be able to get the high loudness levels if the room is large. Passive systems can only be optimized so much.
Once you start needing multiple cabinets, active crossovers become necessary.
To get good coverage of like-frequencies, you want to stack like-drivers. This prevents using passive boxes since each one contains (at least) a high-frequency driver and a low-frequency driver. It’s easiest to put together a sound system when each cabinet covers only one frequency range.
For instance, for a nice sounding 3-way system, you would have low-frequency boxes (the big ones), then medium-sized mid-frequency boxes and finally the smaller high-frequency boxes. These would be stacked or hung, or both—in some sort of array.
A loudspeaker array is the optimum stacking shape for each set of cabinets to give the best combined coverage and overall sound.
You’ve no doubt seen many different array shapes. There are tall towers, high walls, and all sorts of polyhedrons and arcs. The only efficient way to do this is with active crossovers.
Some smaller systems combine active and passive boxes. Even within a single cabinet it is common to find an active crossover used to separate the low- and mid-frequency drivers, while a built-in passive network is used for the high-frequency driver. This is particularly common for super tweeters operating over the last audio octave.
At the other end, an active crossover often is used to add a subwoofer to a passive 2-way system. All combinations are used, but each time a passive crossover shows up, it comes with problems.
One of these is power loss. Passive networks waste valuable power. The extra power needed to make the drivers louder, instead boils off the components and comes out of the box as heat—not sound. Therefore, passive units make you buy a bigger amp.
A couple of additional passive network problems has to do with their impedance.
Impedance restricts power transfer; it’s like resistance, only frequency sensitive.
In order for the passive network to work exactly right, the source impedance (the amplifier’s output plus the wiring impedance) must be as close to zero as possible and not frequency-dependent, and the load impedance (the loudspeaker’s characteristics) must be fixed and not frequency-dependent (sorry, not in this universe; only on Star Trek).
Since these things are not possible, the passive network must be (at best), a simplified and compromised solution to a very complex problem. Consequently, the crossover’s behavior changes with frequency—not something you want for a good sounding system.
One last thing to make matters worse. There is something called back-emf (back-electromotive force: literally, back-voltage) which further contributes to poor sounding speaker systems.
This is the phenomena where, after the signal stops, the speaker cone continues moving, causing the voice coil to move through the magnetic field (now acting like a microphone), creating a new voltage that tries to drive the cable back to the amplifier’s output! If the speaker is allowed to do this, the cone flops around like a dying fish. It does not sound good!
The only way to stop back-emf is to make the loudspeaker “see” a dead short, i.e., zero ohms looking backward, or as close to it as possible—something that’s not gonna happen with a passive network slung between it and the power amp.
All this, and not to mention that inductors saturate at high signal levels causing distortion—another reason you can’t get enough loudness. Or the additional weight and bulk caused by the large inductors required for good low frequency response. Or that it is almost impossible to get high-quality steep slopes passively, so the response suffers.
Or that inductors are way too good at picking up local radio, TV, emergency, and cellular broadcasts, and joyfully mixing them into your audio.
Such is life with passive loudspeaker systems.
Active crossover networks require a power supply to operate and come packaged in single-space, rack-mount units or more often in recent years, built into loudspeakers with power amplifiers.
Looking at the accompanying diagram shows how active crossovers differ from their passive cousins.
For a 2-way system instead of one power amp, you now have two, but they can be smaller for the same loudness level. How much smaller depends on the sensitivity rating of the drivers.
Likewise a 3-way system requires three power amps. You also see and hear the terms bi-amped, and tri-amped applied to 2- and 3-way systems.
Active crossovers cure many ills of the passive systems. Since the crossover filters themselves are safely tucked away inside their own box, away from the driving and loading impedance problems plaguing passive units, they can be made to operate in an almost mathematically perfect manner.
Extremely steep, smooth and well-behaved crossover slopes are easily achieved by active circuitry.
Above, active 2-way crossover, and below, active 3-way crossover.
There are no amplifier power loss problems, since active circuits operate from their own low voltage power supplies. And with the inefficiencies of the passive network removed, the power amps more easily achieve the loudness levels required.
Loudspeaker jitters and tremors caused by inadequately damped back-emf all but disappear once the passive network is removed.
What remains is the amplifier’s inherent output impedance and that of the connecting wire. Here’s where the term damping factor comes up. [Note that the word is damp-ing, not damp-ning as is so often heard; impress your friends.] Damping is a measure of a system’s ability to control the motion of the loudspeaker cone after the signal disappears. No more dying fish.
Siegfried & Russ
Active crossovers go by many names. First, they are either 2-way or 3-way (or even 4-way and 5-way). Then there is the slope rate and order: 24 dB/oct (4th-order), or 18 dB/oct (3rd-order), and so on.
And finally there is a name for the kind of design. The two most common being Linkwitz-Riley and Butterworth, named after Siegfried Linkwitz and Russ Riley who first proposed this application, and Stephen Butterworth who first described the response in 1930.
Up until the mid `80s, the 3rd-order (18 dB/oct) Butterworth design dominated, but still had some problems. Since then, the development (pioneered by Rane and Sundholm) of the 4th-order (24 dB/oct) Linkwitz-Riley design solved these problems, and today is the norm.
What this adds up to is active crossovers are the rule. Luckily, the hardest thing about an active crossover is getting the money to buy one.
After that, most of the work is already done for you. At the most basic level all you really need from an active crossover are two things: to let you set the correct crossover point, and to let you balance driver levels. That’s all.
The first is done by consulting the loudspeaker manufacturer’s data sheet, and dialing it in on the front panel. (That’s assuming a complete factory-made 2-way loudspeaker cabinet, for example. If the box is homemade, then both drivers must be carefully selected so they have the same crossover frequency, otherwise a severe response problem can result.)
Balancing levels is necessary because high frequency drivers are more efficient than low frequency drivers. This means that if you put the same amount of power into each driver, one will sound louder than the other. The one that is the most efficient plays louder. Several methods to balance drivers are always outlined in any good owner’s manual.
Also see Exposing Mythology About Equalizers & Equalization by Dennis Bohn.
Dennis Bohn is a principal partner and vice president of research & development at Rane Corporation. He holds BSEE and MSEE degrees from the University of California at Berkeley. Prior to Rane, he worked as engineering manager for Phase Linear Corporation and as audio application engineer at National Semiconductor Corporation. Bohn is a Fellow of the AES, holds two U.S. patents, is listed in Who’s Who In America and authored the entry on “Equalizers” for the McGraw-Hill Encyclopedia of Science & Technology, 7th edition.
Friday, July 15, 2016
Factors Of A Good Sound Reinforcement System
EDITOR’S NOTE: This fine article was featured in the March 2004 issue of Live Sound International. We reprint it here in celebration of our 25th anniversary.
How many sound systems are in use? Many millions, for sure, and they’re found in all types of venues and for all kinds of programs.
So one would think we’d know exactly how to do it by now. But there seems to be plenty of examples to prove that we don’t. Why should this be? What is it we don’t yet understand? Do we even know enough to know what we don’t know?
Perhaps we should start by trying to define the characteristics of a good system. Not just “it sounds good” but what—exactly—makes the difference between “good” sound and not so good. Then we might be able to quantify how good each characteristic needs to be and how to judge whether it’s good enough or not.
After more than 40 years spent designing and testing sound systems, I’ve finally come up with a list of the factors that I feel make up what we could call quality in a system, and why. In this installment, I’m going to confine my list and discussion to systems for speech reinforcement only. Next time, we’ll look at factors for music systems.
The most important quality factor has to be reliability. No matter how good the performance of a system may be, if it fails to work, it is useless.
Reliability is largely an engineering matter, involving component selection, configuration design, and assembly and installation correctness, for example, but any system can be abused to the point of failure. Significantly, failure may not be abrupt and catastrophic, but instead may take the form of performance decline due to damage.
One particular, and common, example of damage-induced deterioration can be found commonly-used transducer for higher audio frequencies, the horn and compression driver combination. Drivers have a severe amplitude limit; if over driven, the driver diaphragm will impact the phasing plug, an essential part of the structure. If the diaphragm material is metallic, it can fracture and fail.
Some diaphragms, however, are made of a resin-impregnated fabric, which is much less brittle and, therefore, more able to survive a collision with the phasing plug. Repeated collisions, however, still cause progressive deformation (or warping) of the diaphragm, resulting in eventual failure and therefore, progressive decline of the driver’s performance characteristics.
Predicting and detecting this impending failure, however, is not easy to do. The audible change in performance is fairly subtle and can be detected reliably only by careful comparison of the sound of a single questionable driver with that of a known good one. In the field, such a comparison is usually impractical.
Further, a driver that has been used heavily for some time will also exhibit some performance deterioration, even though it has never been over driven into diaphragm collision.
Figure 1 illustrates these performance differences. The frequency response (amplitude versus frequency) of three drivers of the same model (with an impregnated-fabric diaphragm), one new, one well used but apparently undamaged, and one with observable damage.
Figure 1: Measurement of output acoustical and input electrical impedance characteristics of three high-frequency horn drivers of identical model but different usage histories (click to enlarge)
It can be seen that the response at higher frequencies changes with use or abuse. The differences between the upper two measurements are slight, while the third one is significantly different.
There seems to be a good relationship between the measured and (subjectively) observed performances in cases like these, but no real study of this relationship has been performed.
So it would seem that a response measurement could be a valid substitute for a listening test. In fact, such a relationship has been established under certain circumstances, but not definitively in a sound reinforcement context. An investigation of this relationship would certainly be worthwhile.
However, there is another measurement that is easy to make, even though it’s seldom done. The bottom three curves on Figure 1 represent the measured electrical impedance at the input terminals of each of the three drivers. Such a measurement is usually quite easy to make, even on a driver installed in a system.
It’s apparent that these curves separate the characteristics of the three drivers as well as any other common measurement does, especially in the case of the damaged unit, and much more easily. In fact, automated tests of this type have been designed into integrated systems as performance and reliability checks, with good results.
Thus it appears that different types of tests on the same items can yield corresponding results. In fact, experience has shown that such relationships hold in some cases but not in others, and that it may be difficult to predict which is which.
And in many cases, no acceptable substitute for a listening test has yet been found. Worse, some widely accepted tests might prove inadequate.
It’s obvious that any sound system must provide enough sound level at the audience locations to ensure a satisfactory listening experience. Defining what this level actually should be is less obvious, and use of a valid measurement technique is not obvious at all.
Subjective opinions on appropriate sound levels often vary widely as well, depending on a host of factors. (Investigating this matter alone could become a major research project!)
In fact, the correct sound level may not be just a matter of loudness. How well speech is understood (intelligibility) is often the overriding concern, and this is the result of many factors other than just loudness. In some cases, the loudness may need to be set other than as would normally be expected, because of adverse acoustical or system functional characteristics.It may also be found that the audience prefers a sound level different from that which exists near the performer.
Other acoustical factors may be highly significant as well The level of the reinforced sound must be sufficiently higher than that of any background noise so that speech intelligibility or program enjoyment is maintained.
Some desire the added distortion that can result from tube-based components. But is this a bad thing? (click to enlarge)
Some guidelines in this regard have been established empirically, and they may be adequate for most situations. A common and complicating factor is that background noise level may vary significantly, rapidly and unpredictably. Further, since adequate performance in this area may be a matter of life safety, accuracy can be quite important.
It’s often the case that the desired sound level is greater than that which the system is capable of producing without difficulty. This difficulty is the result of one or more components overloading, which results in an audible distortion of the sound.
Distortion may take various forms, depending on the type of component that is overloaded, the magnitude of the overload, and the nature of the program material, among other factors. Therefore, the audibility of the distortion may vary greatly with the situation, and each type of distortion must be evaluated individually.
Many listeners even believe that certain types of distortion are desirable, such as that typically produced by vacuum tube amplifiers. This usually applies to music playback systems in small rooms, however, so it’s unclear if such an effect is valid in a larger sound reinforcement situation.
Some devices are available that deliberately introduce controlled distortion, specifically for pro audio applications. Many have noticed that a limited amount of distortion adds to the apparent loudness of amplified sound, and without being objectionable. If anyone has actually studied this effect, the results remain obscure
The overall timbre, or tonal balance, of a sound system undoubtedly has the strongest influence on the overall perceived quality.This characteristic is easy to measure, both subjectively and objectively, and there is a very good correlation between the two in a small-room configuration. In a large-room sound reinforcement situation, however, this correlation does not hold.
If the system has an overall response that is measurably flat (has nearly the same input-to-output level ratio at all frequencies), it will sound too bright, with the high frequencies being too loud.
A system which sounds subjectively flat, so that the reproduced sound is perceived as being a close duplicate of the source, will have a measured response which rolls down at high frequencies.
Should the analysis be done with a swept filter, which yields more information, or is a stepped filter technique acceptable? What amplitude smoothing or averaging is appropriate? If measurements are taken at single, discrete frequencies, as are commonly done with contemporary techniques, how many measurement points are needed and at what spacing? This could be a major source of misleading data, especially at lower frequencies.
Whatever the technique, how many measurement locations should be taken, and where should they be located? And exactly how should the individual measurements be averaged to yield the overall system response? Also, how much variation between individual measurements is acceptable, and what should be done if the variation exceeds this tolerance?
Schulein documented this discrepancy in 1975 in an elegant experiment and offered a plausible explanation. He noted that in all rooms, the listener receives sound directly from the source and also reflected from the room surfaces.
Here’s the scoop:
Two identical loudspeakers are arranged in a room so that the listener receives the direct sound from both, one close and one distant.
The bandpass filter is first set at 1000 Hz and the attenuator adjusted so that the sound level appears to be the same when the noise is switched from one loudspeaker to the other.
Then the filter is changed to another band and the corresponding band on the equalizer is adjusted until the apparent loudness is the same from each loudspeaker.
This is continued through all bands, and repeated as necessary until no further adjustments are necessary.
Finally, the overall frequency response of the equalizer is measured, this being the correction necessary to make the distant loudspeaker sound like the near one in this room.
In a small room, the level of the direct sound is almost always higher than that of the reflected sound and, therefore, dominates in the perception process.
Because of directional characteristics of human hearing at high frequencies, largely due to head shadowing effects, less total sound energy enters the ears at high frequencies than at lower. This imbalance is perceived as normal.
In a large room with typical acoustics, however, the opposite is true; the level of the reflected, or reverberant, sound is significantly higher than that of the direct at most listener locations. Since this reverberant sound arrives at the listener from all directions rather than just one, more of it enters the ears at high frequencies. Thus the highs are perceived as being louder.
A simple experiment tends to confirm this theory. A loudspeaker is located at head level in a relatively non-reverberant environment and fed with broadband noise. A listener stands one to two meters (about three to six feet) in front of the loudspeaker and slowly turns around while listening to the tonal character of the noise. Typically, the overall tonal balance will change little, if at all, with head direction.
However, if two identical loudspeakers are placed two or three meters apart facing each other and both are fed the same broadband noise, a listener between them, turning around as before, will hear the high frequencies more loudly when his ears are toward the loudspeakers than when he is facing one or the other loudspeaker.
The measured response (and perceived timbre) of a loudspeaker in a room deviates significantly from its performance in an anechoic environment, in ways that are complex and quite difficult to predict.
Also, these deviations are different at each location in the room. Therefore, the only practical solution is to measure the actual response of the completed system and correct it as needed with additional circuitry.
This turns out to be a bit trickier than one might expect, however. If a pure tone, slowly swept in frequency, is fed over a sound system and the resulting level is measured at a point in the audience area, it will be found to consist of strong peaks and valleys, tens of decibels in amplitude, and spaced at intervals of about 1 Hz, caused by room resonances.
It’s almost impossible to get meaningful information from such readings. Besides, we don’t perceive these variations because they are averaged by our hearing process in ways that are only partly understood. The measurements must incorporate averaging which simulates the hearing process.
However, this presents us with a shopping list of unanswered questions pertaining to the measurement techniques. What frequency resolution (bandwidth) is needed?
A first assumption might be to use a bandwidth similar to that of the auditory (critical bandwidth) filters, but system measurements are typically done with third-octave filters, which are considerably wider than critical over much of the spectrum.
Should the analysis be done with a swept filter, which yields more information, or is a stepped filter technique acceptable? What amplitude smoothing or averaging is appropriate? How many measurement locations should be taken, and where should they be located? And exactly how should the individual measurements be averaged to yield the overall system response?
Despite countless practical field experiments in this area, beginning at least 65 years ago, little critical research has been carried out. As a result, there exist only a few de facto standards, and the actual results of these procedures vary considerably in quality.
In addition to the these considerations, it might be expected that nonlinear distortion in any of the system’s components, especially the loudspeakers, would significantly affect its timbre, but such does not seem to be the case. The distortion levels of modern components, properly used, are low enough to be unnoticeable in a reinforcement situation.
As the name suggests, intelligibility is the measure of how easy or difficult it is to understand speech over a system. It’s ultimately measured subjectively and directly, typically using rhyming words as the test signal.
The execution of this test is tedious and time-consuming with only one test subject, which is quite inadequate. Different subjects will render somewhat different results even under apparently identical conditions, and conditions vary significantly with location, program sound levels, room noise, hearing acuity, and many other factors.
Tools like Rational Acoustics Smaart and Meyer SIM can be of great help along with an entire understanding of what’s happening with a system and room? (click to enlarge)
The typically broad variance of test results makes it difficult to determine whether a system is actually performing acceptably or not. It hardly seems worth the rather considerable effort required to execute such a test, but there may be little choice.
Because of these difficulties, a lot of effort has gone into devising an objective test regime, with several products resulting. All involve dedicated gear and techniques, which, while not simple, are quite preferable to subjective tests.
These objective tests have been demonstrated to produce results comparable to those obtained subjectively in some, but not all, conditions. Unfortunately, the worst correlations tend to occur in conditions that produce low scores, exactly where accurate results are most desired. In fact, after extensive experience with all the commonly used objective techniques, Mapp has concluded that all are inadequate.
More Physical Approach
It gets worse. Low intelligibility scores, which indicate serious problems, usually provide little or no information on the nature of these problems. Sometimes one or more physical problems are apparent in such cases, but are these really the causes of the poor performance?
Often, the only way to be sure is to correct the problems and see if that improves the scores. Of course, this may be completely impractical, and in fact, there may be multiple problems, some masking others, so that correcting the most obvious might accomplish nothing useful.
A much more practical approach might be to identify exactly which physical factors adversely affect speech intelligibility, and how, and calibrate physical measurements to subjective effects.
If this were accomplished, then not only would meaningful test methods be available, but effective design criteria could be established to predict results and avoid problems in the design stage. Some significant work has already been done in this area, with results pointing to the ratio of direct to reflected (or reverberant) sound being the most important factor.
Think this over, work on your own list, and next time, we’ll look at key quality factors of music reinforcement systems. (You can read the follow-up article here.)
At the time this article was published in LSI, Bob Thurmond served as principal consultant with G. R. Thurmond and Associates of Austin, TX. Also note that it was originally presented as a paper at the 146th Meeting of the Acoustical Society of America.
Thursday, July 14, 2016
API Announces New 512v Microphone Preamp
Incorporates every feature of the 512c including API's proprietary transformers, 2520 op amp, and a variable output level control.
In response to market demand for built-in signal attenuation, API announces the new 512v microphone preamp.
Incorporating every feature of the 512c, including API’s proprietary transformers and 2520 op amp, the new 512v also includes a variable output level control designed to meet the needs of audio professionals using DAW’s or other input level-sensitive devices in their workflow.
The 512v offers 65 dB of gain as a mic preamp, 45 dB of gain as a line/instrument preamp, and a 20 dB pad switch that can be applied to the incoming signal (whether mic, line or instrument). The 512v also features a 3:1 output transformer tap switch that produces a lower output level, allowing the user the option of driving the transformer harder to achieve higher levels of saturation.
Also included are front and back panel mic input access, front panel combo-style XLR + 1/4-inch line/instrument input, an LED VU meter for gain level indication, 48v Phantom power switch, all built around the traditional API circuit design.
The 512v effectively eliminates the need for third party attenuators, and produces the same analog signal that audio professionals expect from API. Like all API products, the 512v features API’s original 5 year warranty.
The 512v has an MSRP of $995. Units are now shipping.
Posted by House Editor on 07/14 at 01:24 PM
Monday, July 11, 2016
Schertler Now Shipping ARTHUR Format48 Modular Mixer
Swiss manufacturer announces release of new self-configurable mixer created from a choice of different Class-A input and output modules.
Schertler Group announces the release of ARTHUR (Format48) modular mixer, designed for studio or live use, which can be configured and built by the user.
The mixer is created from a choice of different Class-A input and output modules - mic input, mic input ultra low noise, and mic input x4 units, yellow instrument input unit, stereo input unit, L/R master, aux master and external power-in units. These can be combined in any order and quantity to form both compact and large-scale mixer configurations.
The number of units that can be included depends on the power supply used. For simpler combinations of 12 or 25 units, there is a choice of two compact power supplies. A further high-end power supply is also available: This can be used with any combination of units, from just a few to 70 or more. (Power-In units are also required for larger configurations.)
Modules can be added, removed or re-ordered at any time, making ARTHUR one of the most flexible mixers on the market. Further modules are planned for release over the coming months.
ARTHUR’s electronic design offers a complete absence of negative feedback (NFB) from input to output. All filters and summing amps are free from restricting back loops in the mixer’s straightforward design, resulting in a fast response and natural attack. All circuits are built using discrete Class-A components and pure high-voltage DC-amps (without any capacitors in the signal path), offering 30dB headroom and low noise, as well as stability, warmth and transparency.
Configuring and ordering an ARTHUR mixer can be done through the Schertler website (Mixer area). Alongside the technical information and specs for all available modules and accessories, a Configurator enables the various units to be selected in their required quantities. The mixer’s “virtual build” can be previewed along the way as different components are added or removed. External components such as power supplies, side panels and accessories are not supplied as standard and must be ordered separately.
Once delivered, physical assembly of modules is a straightforward process involving a series of connecting rods and hexagonal screws. Users have total freedom to design their own personal channel sequence, as there are no mechanical or electrical restrictions, with the exception of ensuring that the appropriate power supplies have been purchased to support the number of units being used.
ARTHUR is available for pre-order from the Schertler online store (Europe, USA and Canada only) and can also be purchased via Schertler showrooms and distributors worldwide.
Posted by House Editor on 07/11 at 09:11 AM
Tuesday, July 05, 2016
Jake Sinclair Selects Manley ELOP+
Producer for Sia, Weezer, and Fall Out Boy chooses the ELOP+ stereo electro-optical tube compressor/limiter in the studio.
GRAMMY-nominated producer/songwriter/multi-instrumentalist Jake Sinclair recently produced new records for Sia, Weezer, and Fall Out Boy, while simultaneously crafting his side project, Alohaha.
Sinclair’s list of credits includes such artists as Taylor Swift, P!nk, Train, and 5 Seconds of Summer. He runs his studio off a laptop in a room filled with instruments, a few preamps, and his prized Manley ELOP+ stereo electro-optical tube compressor/limiter and not much else.
The ELOP+ is a completely re-engineered version of Manley’s coveted early 1990s-vintage ELOP stereo electro-optical limiter. The ELOP+ uses the company’s highest-performance tube line amplifier and White Follower output stage and relies on a high-voltage switching power supply designed specifically for Manley vacuum tube audio circuits. The “+” also adds a 3:1 compression ratio for greater versatility.
Sinclair wires his two-channel ELOP+ as two separate compressors. One side is the last piece in his vocal chain. The other? “I use the other channel for bass as a tracking compressor, and I go pretty squishy with it,” he explains.
While he respects the old classic optical compressors, to Sinclair’s ears, most of the newer models leave something to be desired-except the ELOP+.
“I’ve tried all of the opto-style compressors,” Sinclair affirms, “and the Manley is by far my favorite. It’s every bit as good as the classic compressors. With a lot of newer opto compressors, as you get past 3 to 5 dB of compression, you start to lose some top end, and it suffocates the sound. But I can put the ELOP+ in Limit mode, keep the filter off, and get a good 5 to 7 dB of compression without losing any high-end tone.”
That’s especially important because Sinclair doesn’t go easy on the compression. “I’m a really guilty overcompressor,” he laughs. “It’s part of the fun! It’s an effect and it also changes the way singers perform. When they’re singing, I apply the compression I intend to use in the final mix. I’d rather track it the way it’s going to be because the singer can react to it, and you get a different performance.”
To do that, it would seem, you need a compressor that can sound transparent. “Yeah,” Sinclair agrees, “and you get that signature sound that we’ve heard on millions of records; the ELOP+ just nails it.” The result, he says, is an airy vocal that sounds like it’s supposed to. “EveAnna Manley described the ELOP+ as just a volume knob,” he muses, “but it’s more than that for me. There’s some kind of tube-y glueness.”
Sinclair has reduced his use of tube devices in recent years but you’ll never get his ELOP+ away from him. “Except for microphones, the ELOP+ is the only piece of tube equipment I still have,” he admits. “Tube preamps aren’t for me. But I will always have an ELOP+.”
Posted by House Editor on 07/05 at 08:03 AM
Tuesday, June 28, 2016
Suffolk New College Adds Audient For Music Technology Course
ASP4816 analog console installed as part of a studio upgrade at college in Suffolk, United Kingdom.
“My background is in the studio so I can appreciate good workmanship,” says Suffolk New College Music Technology Course leader, Craig Shimmon, who’s very pleased with the arrival of the new Audient ASP4816 desk. “The other lecturers have commented on the superior build quality and the clean preamps; the layout is a dream to teach with,” he adds.
Installed as part of an upgrade to the studio at the college, the analog mixing console forms an integral part of the Music Technology course, teaching Level 2 and two years of Level 3 Diploma students.
Shimmon explains, “Our department consists of industry professional engineers, composers, producers and DJs teaching a range of different Music Technology specialisms, with industry-standard equipment and facilities. All in all approximately 60 Music Technology students will have hands-on access to the studio, with a further 60 music performance students benefiting from being recorded.
With all the key features of a large console in a compact and ergonomic form, the ASP4816 also features in-line architecture, ideal for teaching signal flow, which fits perfectly at Suffolk New College.
“I love the in-line faders,” enthuses Shimmon. “With previous desks we have had to use Mix B pots or split the desk which can be confusing. Teaching with the Audient console just makes it a whole lot easier. This coupled with the ability to route different channels to different tape outs makes this a very powerful machine.” Which is exactly how New Suffolk College meets its strategic aim, “to provide high quality learning and teaching, and gain recognition as an excellent and innovative provider.”
Audient has every faith that with the new, improved set up and under Craig Shimmon’s instruction, New Suffolk College music technology students will go far.
Suffolk New College
Posted by House Editor on 06/28 at 09:19 AM
Thursday, June 23, 2016
Smalltown America Studio Captures Live Performances With Audient
Northern Irish facility showcases live bands from across the UK and Ireland with its series of ‘Live @ STA’ events.
Smalltown America Studio (STA) in Derry, showcases acts from across the UK and Ireland with its series of ‘Live @ STA’ events.
“The bands play live on the floor in our 1600sq foott live room,” confirms studio engineer, Caolán Austin. “We run our lines through our (Audient) ASP8024 then to ProTools; we send the mixed feed from the desk to an encoder box and stream the audio and video on our website. People can watch at home if they can’t make it to the gig.
“Fitting bands, PA and a crowd into the space is quite a feat,” he laughs, explaining that the small audience usually numbers about 50. It’s not a problem if you missed it – or want to relive the experience, either. “After the show we mix everything and each show is released as an album.”
Austin describes the 36-channel Audient desk as “beautiful” and “the focal point of our whole studio – and quite a magnificent one, at that,” taking pride in the Northern Irish facility. “For traditional recording sessions, our tracking/monitoring workflow has been entirely designed around the analog desk.
“As we have such a spacious live room, most bands want to come in and track live. We have devised a system of ‘stations’ within the studio. Drums are permanently set up, we have an amplifier room to send signals to, there is a vocalist station and so on. This reduces initial set up time to around 30 minutes. Bands on a budget want to get working straight away; using this method, we can get great results quickly. It wouldn’t be possible without the ASP8024’s flexible routing and the use of TFS for line/insert points for patching our outboard presets.
“For monitoring, we run a Behringer Powerplay headphone system with P-16 modules. Each auxiliary bus on the desk is wired to correspond with a headphone channel, including control room foldback. During a session, we can route new audio paths quickly, while still giving bands a flexible channel count to ensure headphone mixes are good as they can be,” adds Austin, continuously mindful that some clients need to watch their pennies – but not scrimping on quality. “All of this ensures our workflow is still incredibly quick and efficient.”
Smalltown America Studio