Wednesday, April 10, 2013
Allen & Heath Unveils Qu-16 Digital Rack Mount Mixer At Prolight+Sound 2013
Compact unit incorporates technologies pioneered in company's GLD and iLive digital mixing systems
Allen & Heath has announced the new Qu-16 rack-mountable digital mixer, incorporating technologies pioneered in the company’s GLD and iLive digital mixing systems.
An overview of the facilities of the new Qu-16:
—16 Mono Inputs (TRS + XLR)
—3 Stereo Inputs (TRS)
— 4 Stereo FX Returns
—12 Mix Outputs (LR, Mono Mix 1-4, Stereo Mix 1-3)
—4 FX Engines
—AES Digital Out
—19-in Rack Mountable
—Recallable AnaLOGIQ Preamps
—Qu-Drive direct Multitrack Recording / Playback on USB drives
—800 x 480 Touchscreen
—iLive FX Library
—dSNAKE Remote Audio Port
—Compatible with ME Personal Mixing System
—Qu-Pad iPad App
—USB Audio Streaming
—DAW MIDI Control
—4 Mute Groups
—Trim, polarity, HPF, gate, insert, 4-band PEQ, compressor and delay on all Inputs
—Insert, 1/3-octave GEQ, compressor and delay on Main LR and Mono Mix outputs
—Insert, 4-band PEQ, compressor and delay on Stereo Mix outputs
—Built-in Signal Generator
—RTA with Peak Band indication
The Qu-16’s sixteen AnalogiQ total recall preamps feature zero crossing detection and a padless 1-dB-step gain stage, closely allied to the DSP for optimal gain accuracy and audio transparency. The analog signal is captured by low-latency 24-bit analog-to-digital converters matched to high-quality 24-bit digital-to-analog converters to deliver the required outputs.
The 800 x 480, 16-million-color Touchscreen and its dedicated data encoder form the heart of the Qu-16 interface, providing super-fast, easy access to all settings. The user-friendly interface has been designed with clarity. Dedicated keys and screen tabs quickly guide the user to meter and RTA views, FX racks, channel processing, USB audio control, scenes, setup menus and much more.
All key processing tools are presented in a clean layout on the SuperStrip, with one function per physical control. The SuperStrip is complemented by an onscreen Touch Channel for intuitive access to full processing parameters without clutter or complex menu structures.
Processing for Mono and Stereo inputs includes trim, polarity, HPF, gate, insert, 4 band PEQ, compressor and delay. The main LR and the Mono mixes have controls for Insert, 1/3-octave GEQ, compressor and delay. The Stereo mixes provide Insert, 4-band PEQ, compressor, delay and balance control.
Fader automation is essential for rapid mixing, especially when dealing with multiple monitor mixes. Qu-16 offers 17 motorized ALPS faders, 16 arranged over two layers, allowing instant access to all channels and masters in a compact space, plus a dedicated master fader which dynamically follows the mix selection. A third custom layer is available for ad-hoc user strip layout, where any combination of Inputs, FX Sends, FX Returns and Mix masters can be assigned.
Qu-16’s dynamics and FX algorithms are derived from the FX used in the iLive pro touring series. Qu-16 offers four stereo iLive FX engines, featuring carefully crafted emulations of legendary classic reverbs, gated reverbs, delays, modulators, flangers and more. The FX library has the ability to grow with future firmware releases. FX are returned to the mix on dedicated return channels, thus not tying up mono and stereo input channels. Each Stereo FX Return has a dedicated 4-band PEQ.
The Qu-16 can store up to 100 full scenes for recall at will. Channels and mixes can be made safe from scene recall. For example, if an instrument or mic gets swapped out after the soundcheck, the channel can be made safe to avoid settings being overridden by Scene recalls. Or if a broadcast feed or walk-in iPod is added last-minute before the show kicks off, that mix or channel can be made safe from any scene change.
In addition, single parameter updates can be blocked using per scene Recall Filters or a Global Recall Filter. So if you tweak the graphic EQ to reflect the room response when the audience gets in, you can block this to prevent any overwriting at scene change.
Custom settings for each EQ, compressor, channel or FX can be saved as Library presets. This lets the user, for example store “tried and tested” SM58 EQ or reverb pattern and apply it to other channels or shows. Libraries, Scenes and the complete Show configuration can be saved to a USB key, carried to the next show, ready to use on another Qu-16.
Qu-16’s built-in interface streams multitrack audio to Mac channel 1 to 16, the Main LR mix, and three selectable stereo pairs. The returns from the Mac can be assigned to the 16 Mono channels plus stereos.
The interface is class-compliant on Mac OS X, with no need to install a driver. It will be recognized straightaway by any DAW supporting Core Audio, including Logic, Cubase, Reaper, and Pro Tools.
Standard MIDI control is tunneled over the USB connection so you can easily map the faders to the tracks of your favorite DAW. Alternatively, a MIDI driver is available for use with the Ethernet port.
Qu-16 has an integrated multitrack USB recorder, providing 18 channels of 48 kHz, 24-bit recording and playback straight to/from a USB hard drive. A selectable stereo pair can be recorded alongside the 16 Mono channels, and multitrack audio can be played back to the 16 Mono channels plus ST1.
On top of this, Qu-Drive also provides stereo recording, patchable from any pair of Mix outputs, the Main LR (pre, post, or summed to mono) or even the PAFL bus, with 2-track stereo playback to ST3.
Qu-16 is equipped with five cores of high efficiency ARM core processing, with dedicated ARM cores running the touchscreen display and surface, USB streaming, Qu-Drive multi-channel USB recording/playback, Ethernet and fader automation. Between them the ARM cores provide state-of-the-art processing, working in parallel to deliver extensive control, instant-on operation, and very fast response.
The mixer’s DSP farm exploits next generation dual core DSPs, giving 10 DSP cores, with 8 dedicated to the channel and mix processing alone. With so much DSP power under the hood the channel processing is only using a fraction of capacity, so Qu-16 is future-proofed, with ample room for updates and extra functionality.
The Qu-16 DSP architecture employs varied bit depths, tailored to specific algorithms, with 48 bits on critical EQ functions and a 56-bit accumulator on the mix bus where it really counts, allowing every nuance of the audio to be captured in the final mix.
Made from 18 gauge, cold-rolled Zintec steel, Qu-16’s frame is designed for strength, rigidity and ease of rack mounting. The mixer’s sleek profile generates optimal airflow through the mixer, eliminating the need for any fans.
The shape has some unexpected benefits too. The space beneath it is useful for keeping a USB drive, a talkback mic, a cue sheet and other clutter tucked out of the way.
The Qu-Pad iPad app allows adjustment of monitors on stage, as well as roaming around the venue while tweaking the PA, and then mixing a show from the heart of the audience. Qu-Pad connects to the mixer over Wi-Fi and gives instant access to all live mixing parameters and settings. It requires the connection of a Wi-Fi router or access point to the Qu-16 Network port.
The Qu-16 is fully compatible with the Allen & Heath ME personal mixing system. Any number of ME-1 personal mixers can be chained from the dSNAKE port (or from an AR2412 Stagebox if you’ve got one connected to the dSNAKE port). Each performer can be given tailored control over their own mix, leaving the engineer free to focus on the audience experience.
If you’re thinking of trading in the copper multicore for a Cat-5 digital snake, Qu-16’s dSNAKE port provides a solution, allowing connection to a remote AR2412 or AR84 Stagebox. dSNAKE is Allen & Heath’s proprietary networking solution, boasting a transport latency of just 105us over cable runs of up to 120 meters/390 feet.
Allen & Heath
American Music & Sound
Church Sound: Make Up For Lost Instruments By Filling In The Hole
There are two ways of treating missing instrument
The drummer called in sick. The bassist’s car broke down. The guitarist had a family emergency. Whatever the reason, you are now missing an instrument in the band.
It’s time to adjust your mix to fill in the hole.
There are two ways of treating missing instruments:
1) Make it obvious in the mix. For example, if the only electric sound in the band was the electric guitar then mix it as an acoustic set.
2) Cover up the vacancy with other sounds. It’s here that I’m parking for today.
Methods For Filling In The Hole
—Pretend you never had that instrument. This will require denying knowledge of the musician and their family. This is not a method I recommend. It simply gets too complicated and if you accidentally mention their name a few years later, then it all falls apart.
—Look at alternative miking for bringing in the missing frequencies. A djembe is a great percussion instrument with the slap on the top skin. If the drummer calls in sick, add a mic to the bottom of the djembe to bring in more low-end frequencies into your mix.
—Look for areas for boosting. If the pianist couldn’t make it in, look at boosting the upper-mids and highs on the acoustic guitar. No bassist?Boost a bit of the lows in the electric guitar. Your goal isn’t completely filling in the frequency holes, only to make the holes less obvious.
—Look for areas for cutting. Any time an instrument is removed from a mix, the overall balance of the instruments and vocals has changed. Bottom line, it’s time to re-evaluate your mix.
I will note that a missing instrument should give rise to arrangement changes by the worship leader, but that’s not always possible—or in some cases—necessary.
The Take Away
The instruments present at your mid-week practice aren’t guaranteed to be there for the church service. That’s part of live audio production – things change.
The good news is your mix doesn’t have to come crashing down. Consider how the mix sounds without that instrument and start making changes to close up the gap. You shouldn’t try making a guitar sound like a bass and a guitar, but you can make subtle changes that fill in some of the missing low end.
Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians, and can even tell you the signs the sound guy is having a mental breakdown. To view the original article and to make comments, go here.
SSL Launches Its First-Ever Live Audio Mixing Console At Prolight+Sound 2013
Offers 976 inputs and outputs and 192 full processing audio "paths" at 96 kHz
Solid State Logic (SSL) has announced the launch of Live, the company’s first console for live sound production, at this week’s Prolight + Sound 2013 show (Hall 5.1, Stand B.73).
“So many people have been asking us to make a console for live for so long and the time has come,” states Antony David, Managing Director at SSL. “It has been a couple of years in the making because we like to get things right at SSL and we are very confident that we have created a console engineers will fall in love with.
“It is very exciting to be entering a completely new area of the industry, with a new technology platform and a fresh approach to how a live console should sound and how to give engineers a control surface that helps them deliver exceptional performances. We are really looking forward to how the professional live sound community reacts.”
Live is a flexible, powerful digital audio console designed specifically to meet the demands of all aspects of live sound production. Based on SSL’s new Tempest processing platform, Live has plenty of power and it is deployed in a uniquely versatile way. It offers 976 inputs and outputs and 192 full processing audio “paths” at 96 kHz.
How those “paths” are configured is very flexible, with power allocated to channels, auxes, stem groups and masters configured to suit the needs of each show. All processing is built into the console surface, and it has a collection of I/O connectivity built into the frame. A full range of Stagebox I/O connects to the console via MADI with the potential for larger systems to make use of SSL’s own Blacklight technology that carries up to 256 channels of bi-directional audio and control via a single fibre connection.
The control surface combines multiple tablet style multi-gesture touch screens with hardware ergonomics, strong visual feedback and a collection of innovative new features. It allows engineers to work how they prefer, touch screen or classic hardware technology, or to combine them both.
Very high audio quality is a primary design feature, including studio grade SuperAnalogue mic preamps, 24-bit/96 kHz A/D D/A conversion, to 64-bit internal processing and 96 kHz operation throughout. Live also offers a collection of 30 new effects and audio analysis tools, which take SSL’s studio grade processing and re-craft it for live. They also have their own dedicated processing power.
Live is due to ship in September 2013. Depending upon configuration prices will range between:
GBP—£48,000 and £75,000
Euro—€57,500 and €90,000
USD—$84,000 and $130,000
Solid State Logic (SSL)
Tuesday, April 09, 2013
PreSonus Debuts New StudioLive 32.4.2AI Digital Mixing System With Active Integration
More than 64 times the processing power and 10,000 times more RAM than the StudioLive 24.4.2
PreSonus has introduced the StudioLive 32.4.2AI 32-channel performance and recording digital mixer.
This new mixer features next-generation Active Integration technology, including a dual-core computing engine that packs over 64 times the processing power and 10,000 times more RAM than the StudioLive 24.4.2.
Its sophisticated, integral communications also makes possible wireless control of the mixer without requiring an external computer.
The new console includes 32 Class A XMAX mic preamps with individually switched phantom power (plus an XMAX preamp for the Talkback input with always-on phantom), 32 line inputs, 14 aux mixes, 4 subgroups with variable output delay, Fat Channel dynamics processing and parametric EQ, a 48x34 FireWire S800 audio interface, and more.
The Fat Channel is the same as is found on the StudioLive 24.4.2, including routing, panning, a high-pass filter and polarity reverse on every channel, and a full-feature gate, full-featured compressor, limiter, and 4-band fully parametric EQ on every channel, aux, subgroup, and effects bus.
Taking advantage of the Active Integration engine’s considerable processing power, the new mixer allows users to create two complete sets of EQ and dynamics settings for a channel and then make quick A/B comparisons with the Alt EQ/Dyn button.
Also new in this model are 6 mute groups with All On/All Off switches and six user-assignable Quick Scene Recall buttons that lets the user load specified, saved mixer scenes-sort of a speed dial for mixer scenes.
The StudioLive 32.4.2AI also sports four internal effects buses: two with reverb and two with delay effects.
Unlike previous models, the StudioLive 32.4.2AI also has an Ethernet port that allows connection to an existing router-based network with an Ethernet cable or completely wirelessly. A USB 2.0 port hosts the included USB Wi-Fi LAN adapter for situations where a less powerful, ad hoc Wi-Fi network is sufficient.
Using either wireless connection, the mixer can be directly controlled from a laptop or iPad, and the aux mixes can be controlled from an iPhone or iPod touch; a FireWire connection to a computer is not required.
A line of option cards will be available for the StudioLive 32.4.2AI in late 2013. Every option card comes equipped with dual FireWire S800 and S/PDIF stereo output, as well as Ethernet (for control only). Two cards that add Dante for audio networking and Thunderbolt for even faster digital transfers are in development.
StudioLive 32.4.2AI is tightly integrated with updated versions of the same powerful software suite introduced for the earlier StudioLive models: VSL-AI control/editor/librarian for Mac and Windows, SL Remote-AI wireless remote control for iPad, QMix-AI wireless aux-mix control for iPhone/iPod touch, Capture 2.0 preconfigured audio-recording software, and the highly lauded Studio One Artist DAW.
The StudioLive 32.4.2AI digital mixer is expected to be available in April 2013, with an anticipated MSRP of $4,999 and MAP of $3,999.
SPL Debuts New Audio & MADI Interfaces At Prolight+Sound 2013
24-bit/192 kHz-capable Crimson is housed in a desktop footprint yet offers 30 I/O channels (10 recording and 20 playback)
SPL (Sound Performance Lab) has introduced the new Madison 16+16 channel studio I/O and Crimson USB audio interface and controller at this week’s Prolight+Sound 2013 show in Frankfurt.
Crimson combines a USB interface with high-quality pre-amps and a separate, fully-featured monitor controller, creating a cost-effective, portable package. Play, playback, record, convert, control, and listen with a single device.
The 24-bit/192 kHz-capable Crimson is housed in a desktop footprint yet offers 30 I/O channels (10 recording and 20 playback), +/-18-volt operational voltage (for professional levels up to +24 dB), two discrete Class A +/-30-volt high-voltage mic pre-amps, two Hi-Z instrument pre-amps, two separate headphone amps (individually controllable), dual stereo loudspeaker set connections (and control), monitor signal mix function (from playback and recording paths), MIDI input and output, plus USB 2.0 as well as S/PDIF input and output.
With MADI emerging as the new studio I/O standard, the Madison 16+16 channel studio I/O combines conversion, an ultra-reliable clock, and MADI processing with a simple- to-operate, easy-to-expand, and great-sounding system housed in a compact 1U rack-mountable unit offering 16 input and 16 output channels.
The built-in MADI port offers pristine and near- zero-latency digital transmission of up to 64 inputs and outputs with galvanic insulation (via four units connected on a single MADI port).
Madison includes 16 superb-sounding A/D and D/A convertors with an 18-volt analog section (36-volt audio rail for +24 dBu professional level), 44.1 to 192 kHz operation (with ±10 percent varispeed), SPL clock-shop (for jitter-free operation), ultra-compatible MADI I/O (with very low conversion latencies), four directly-selectable reference levels (15/18/22/24), simple operation and setup (directly from front panel controls), distance-readable meters (for clear level status indication of any I/O channel), fan-less silent lower-power design (30 watts max), redundant power supply (optional), and four industry-standard (TASCAM) D-Sub analog 8-channel multicores.
The Crimson USB audio interface and controller (Model 1250) and Madison 16+16 channel studio I/O (Model 1260) will be available for purchase worldwide from SPL dealers with a RRP of €549 EURO (including German VAT)/$725 USD and €1,499 EURO (including German VAT)/$1,899 USD, respectively. Availability is scheduled for June 2013.
Church Sound: Getting Great-Sounding Sermon Podcasts
A step-by-step process to deliver improvement
I listen to a lot of podcasts. Several hours a week, I’m at the gym working out (which is why I’m so buff), listening to a podcast.
One thing that drives me nuts is having to constantly adjust the volume on my iPhone because the level of the podcast is all over the place. I used to listen to a lot of sermons from other churches; some large churches that you would have heard of, others were smaller.
But I stopped after a while because so many of them had terrible audio. The levels were inconsistent, or distorted, or noisy, or there were other issues.
Now, it’s true that many churches have this dialed in. But I get e-mails from people fairly regularly asking for help in getting the sermon sounding good online.
So I figured I’d let you into our process, which I think creates some pretty decent sounding podcasts.
But before we get to the how, let’s consider the what. What do we want to accomplish?
I can’t stand music that is over-compressed, with all the dynamic range taken out (which is why I tend to listen to older music). However, when it comes to podcasts, I really don’t want dynamic range. When I’m huffing and puffing on the elliptical, I don’t want to keep turning the volume up when the pastor gets quiet, and having my ears blown out when he gets loud again.
Here is a typical weekend sermon recording waveform. This is where we start.
Others may disagree with me (and I’m sure we’ll hear from some), but I want to limit my sermon podcast’s dynamic range as much as possible. I’ve found this makes the audio far easier to listen to in the car, on the computer, on a walk or at the gym (which is where people tend to listen to them).
There are many ways to get to a very limited dynamic range, and we’ll talk about them in the next post. But here is something interesting. You might worry about loosing the cues that come from varying levels of speech if the dynamics are squashed.
And this is where we end up. Notice the waveform is almost completely solid. The volume varies very little.
As the pastor softens up and gets quiet to make a poignant statement, you may think it needs to be quieter. However, I’ve found that the tonal qualities of the voice can convey those cues regardless of the level.
And let’s be honest, people are going to be turning the volume up to hear it anyway if they are in the car or working out. So my philosophy is to do the work for them, and keep the volume consistent.
Minimize File Size, Maximize Quality
I try to keep my sermon podcast file sizes down below 15 MB. They download quickly, even over 3G, and don’t take up a ton of room on the MP3 player. To get there, I use the LAME MP3 encoder—which is one of the best available. For a long time, I used VBR (Variable Bit Rate) encoding on my podcasts, with the quality level set to 20 (which equals roughly 48 kbps, average).
However, I recently learned that iTunes has a problem playing VBR files; well technically, it’s a problem pausing them. It seems that pausing a VBR-encoded file will cause iTunes to back up some amount of time before playing back. How much it backs up depends on how far into a program you are. It’s not the end of the world, but it is annoying.
So recently, I’ve switched to CBR (Constant Bit Rate) encoding. After doing some tests, I found it very difficult to distinguish between VBR at 20% and CBR at 48 kbps. There was a tiny bit of difference at the high end, but this is speech we’re talking about, not high quality music.
Both sound more than acceptable on both my UE7s and our NS10 monitor speakers, and I heard no difference through the speakers in my MacBook Pro. So for now, CBR it is.
Side note: It seems that iTunes 10.7 has a better time with VBR files, but I know a lot of people are still using older versions. Given that I don’t hear any penalty in quality, and the file sizes are very comparable, I’m sticking with CBR.
So that’s what we’re going for; minimal dynamic range, low file sizes, high quality.
Now, how do we get there? Like most things, it’s a multi-step process to achieve the best results.
But before we have anything to edit, compress or publish on the interwebs, we must first record something.
Record As Close To The Source As Possible
If you’re working with a digital console and are using a virtual sound check system, you are already in great shape for recording the message. That’s how we do it at Coast Hills, using our RME MADIFace to run the audio directly after the A/D conversion into the MacBook Pro and on into Reaper.
As I wrote recently in Automating Reaper (Again), I record a 2-track board mix of all three services, and a discreet track for our pastor on both Sunday AM services. I use the discreet track for the podcast; the 2-track is backup only.
Since we’re recording speech (for this purpose anyway), I don’t worry about getting to 192 KHz or anything crazy. 48 or 44.1 KHz at 16-24 bits is just fine. Our system runs at 48 KHz, 24 bits, so that’s what we record as a series of WAV files. WAVs are uncompressed, so the quality is quite good (AIFF files would also be a good choice). You want to record uncompressed if at all possible.
If you’re using an analog system, don’t fret. Use the direct outs—or in a pinch, the output of the insert jacks—to come directly out of the pastor’s mic channel to your recorder. You really want to pick off the output before EQ, compression, or other processing.
The reason for this is that most times, you are making EQ and compression adjustments on your console for the room—which is where most people are listening. However, those same settings may not work for the recording, and there are advantages to making some EQ and compression choices specifically for the podcast.
You can record to a CD (we do an archive CD of our 9 am service each week for backup), but I really prefer to record straight to the computer since we’ll be editing and processing the file there anyway. Even an inexpensive USB interface like a Lexicon Alpha (about $65) will get you great sounding direct recording.
Combine that with a laptop, Mac Mini or inexpensive PC and a copy of Reaper and you’re in business. If you’re stuck with a CD, rip it into DAW (Digital Audio Workstation) software for further editing and processing.
Process For The Web
What works in the room may not work online. You will have to do some experimentation here, but I tend to high-pass our pastor fairly high (up around 130-140), and boost the upper mids by 1-2 dB. I’m trying to add a little bit of clarity to make it easier to listen to in loud environments. But be careful here because you can easily make it annoyingly bright.
I suggest you try some settings, encode a section of the sermon and listen to it on several platforms to see how you did. EQ settings that work in your 7506s may not work on a cheap set of computer speakers, so check it out. And don’t forget many people will listen to the podcast with Apple’s cheap, white earbuds.
Since my pastor’s voice is recorded pre-EQ, I do some subtle changes to make him easier to listen to. Then I hit it with the compressors.
Currently, I’m using R-Channel from Waves to do both EQ and compression. I will say the R-Channel compressor is one of the more transparent ones I’ve heard; I routinely have it hitting 12+ dB of gain reduction and it’s really tough to hear.
Before I had that plug in, I achieved results almost as good with a combination of ReaEQ and ReaComp—both plug-ins come with Reaper. Most DAWs have basic EQ and compressors built-in, so play with those first before you go spending money on plug-ins.
However, it would be worth it to sign up for Waves mailing list; they often do super deals on individual plug-ins and you can pick up one or two that will rock pretty cheap.
Limiting For The Win
The final step is to use a mastering limiter to really clamp down the dynamic range. I was using JS: LOSER MasterLimiter (included w/ Reaper) for quite a while along with ArdazMaximzer5.
The MasterLimiter allows you to set a maximum level (I went with -.01 dB) that it will allow; it’s a brick-wall limiter so nothing gets over that. It will also do some compression to keep the signal level up.
The Maximizer does some other magic to raise the overall level without driving it over the limit. That combination worked really well, and sounded pretty good once we got it dialed in.
This is the rendered audio file (in this case, a stereo AIFF for the CD ministry folks). The MP3 is a mono version of the same thing.
Then I picked up the Waves L3 UltraMaximizer. And that was pretty much that. After setting a few sliders, I can pretty much crank the level like crazy and it sounds amazing. I showed this picture last time, but you can see how little variation in the waveform we have on the rendered file, indicating very little dynamic range. If this were music, I would be upset, but for a speech podcast, it’s about perfect.
Rendering To MP3
Most DAWs can render out to an MP3 file. If you have the option to use the LAME encoder (which you have to do in Audacity or Reaper), use it. It’s a great encoder that produces better-sounding MP3s at lower bitrates than most other encoders.
As I noted earlier, I use the 48 kbps CBR setting, and render in mono. I haven’t had any problems with mono files anywhere, and given that it’s spoken word, stereo unnecessarily doubles the file size.
I got to these settings (and all the ones I didn’t tell you about) by doing a lot of experimenting. I intentionally didn’t show you all my EQ, compression and limiting settings because they don’t really matter—they are all specific for our pastor.
If you spend a few hours working on a chunk of the message getting the processing settings right, then tweak your rendering settings, you’ll end up with great results. Then don’t forget to save those settings as presets so you can use them next week.
I’ve mentioned a ton of stuff in this article, but here is a recap of what I recommend for this process. If you have something else that works, by all means, keep using it. If you’re looking for a place to start, consider this list.
Audacity (Free recording/editing software; good basic and free)
Reaper (Full-featured DAW; $60 for non-profit use, incredibly powerful, still easy to use)
Lexicon Alpha (simple 2-track USB audio interface; about $65)
ArdazMaximizer5 (Free maximizing plug-in)
Waves L3 UltraMaximizer (amazing maximizing plug-in; $350, but look for it on sale)
Waves R-Channel (amazing channel strip; $175, but look for it on sale )
If you want to hear the results of this processing, you can check out the Coast HIlls media page. Listen to the MP3 files, those all have processing similar to what we’re taking about here. Keep in mind, since we have most of this stuff in presets, it takes about 5-10 minutes to edit, process, render, upload and post our podcasts.
Once we did the hard work, the weekly stuff is easy.
Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog Church Tech Arts. He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.
Monday, April 08, 2013
South Africa’s Howard Technologies Adds Soundcraft Si Compact 24 Console To Inventory
Console helps meets needs of growing work roster of multi-act shows
A new Soundcraft Si Compact 24 has been added to the inventory of South African sound specialist Howard Technologies, after the company purchased the desk from Bloemfontein-based Toms Music.
Set up by Irvin Howard in the year 2000, the company specializes in all aspects of staging and production, and with a growing work roster of multi-act shows, needed a digital desk that was compatible with this type of presentation, and also expandable.
“We operate in the corporate sector doing conferences and concerts — and memorising each set was becoming a challenge,” Howard states. “We also handle a lot of church conferences.”
Howard, whose past experience with Soundcraft includes the LX7, adds, “Soundcraft has always been one of my favorite desks — they have strong pre-amps and are really clean sounding. In this instance, expandability was important, as was simplicity, portability and the ability to handle multi-track recording through Logic/Pro Tools.”
This wish list had evolved from less accommodating experiences on other branded desks, which he had found less than intuitive. “Then I did a show where the SI Compact 32 was set up. I was really hesitant at first but the guy who had set it up literally took 10 minutes to explain the desk. It was a breeze. I knew then that I wanted a Soundcraft Si Compact.”
Drawing on the heritage of Soundcraft’s digital Si Series, the Si Compact packs high quality mixing facilities and Lexicon effects into a small footprint console. It does so without compromising a feature set that includes 24 recallable mic pre amps plus four stereo returns, AES in and 64 input option slot with a capacity to combine up to 40 of these inputs to mix; each input is equipped with compressors, gates, parametric EQ, high-pass filter and delay.
Howard is delighted to be carrying a brand that has such a high reputation In South Africa. “There is not a band that will complain about being mixed through a Soundcraft desk — and the name appears on most technical riders.
“This desk will now not leave my sight. Either myself or my technicians will always be present, even if we’re not mixing the sound. We’ll leave our other desks to handle the dry hire.”
Soundcraft mixing consoles are distributed in South Africa, along with the entire Harman Professional portfolio, by Wild and Marr.
More Than A Project: Audio For Alan Parsons In The Live Realm
A studio legend takes it out on the road
Few have achieved the level of success of Alan Parsons as an engineer/producer, as well as a performing artist in his own right.
After beginning his career as an assistant at Abbey Road Studios at age 19, he worked on iconic records such as Abbey Road and Let It Be before further cementing his reputation on projects with The Hollies, Al Stewart, Paul McCartney and Wings and, of course, with Pink Floyd as engineer on Dark Side of the Moon.
Forming the Alan Parsons Project with producer/songwriter Eric Woolfson in 1975, Parsons released a string of successful records that garnered him multiple Grammy nominations and enduring critical acclaim.
He remains active in the studio – recently undertaking projects such as The Raven That Refused To Sing (And Other Stories) by Porcupine Tree’s Steven Wilson, and Jake Shimabukuro’s Grand Ukulele, and he also continues to tour with the Alan Parsons Live Project (APLP), completing a rare run of U.S. shows in February, 2013 before heading to Europe for more live dates.
Given Parsons’ reputation as a recording artist and engineer, it’s natural for anyone mixing front of house for APLP to have a case of the nerves the first time out. “I’d be lying if I said I wasn’t nervous at all,” says Nashville-based Martin Frey, who’s been the band’s house engineer since 2010 and is a long-time fan of Parsons’ music. “But I did my homework. I still have those records on vinyl, and when I got the set list I arranged the tracks in order on my computer, listened to them for several weeks and made notes, so I went in with a clear mental picture of what I was going for.”
Frey adds that Parsons immediately put him at ease prior to the first show: “Alan came out into the house to listen during sound check so I went over and said, ‘Well Mr. Parsons, what do you think?’ And he replied, ‘could you push the vocals back in the mix and put a little more bottom on the snare? And by the way, please call me Alan.’ He’s just a straightahead guy and knows exactly what he wants to hear.”
Front of house engineer Martin Frey with Alan Parsons prior to a show on the recent tour. (click to enlarge)
For his part, Parsons is content to leave the mix to Frey. There are audience members, however – audio engineers and enthusiasts of all descriptions who are fans of Parsons’ work – that are quite willing to offer comments, Frey says, but, happily, this feedback has been positive.
While audio professionals have long been attracted to Parsons’ work and APLP shows, interest has increased since he released a series of videos entitled the Art and Science of Sound Recording in 2010, which trace the development of sound recording “from Edison to the MP3.”
Speaking with me before heading off for the European leg of the tour, Parsons said that the intensified interest is encouraging, adding that he’s also considering a follow-up project that will deal with live sound exclusively.
Pick A Console
Frey’s approach to mixing APLP is similar to that he employs with other artists he’s worked with, which range widely and includes rock bands such as Journey, country stars Tanya Tucker and Big & Rich, and large orchestras. “Although I work in a wide variety of musical genres, from a mixing and production point of view, I approach everything the same way,” he notes. “I study the material, make notes, and unless someone tells me differently, I go for what the records sound like.”
The Alan Parsons Live Project in concert. Photo credit: Chezaray.com. (click to enlarge)
When tuning the PA in advance of shows on the recent tour, Frey utilized Sound Check, a test disc released in 1993 by Parsons and Stephen Court. “I’ve been using that for 20 years to listen to the human voice, which allows me to ascertain the intelligibility of the PA immediately, and to the drums and bass to fine-tune the low end,” he explains.
That’s a key concern, particularly since for the most part, APLP doesn’t carry sound (or lights), which remained the case with this tour. “Even if we did,” Parsons says, “we’re still at the mercy of the acoustics of the venue. We try to be as high fidelity a live act as possible. It’s not always easy, but I try to make sure we have the best engineers possible who will achieve the best results they can with the equipment they have.”
During the U.S. run, Frey mixed on six different consoles – a Midas Heritage 3000 analog, as well as Yamaha CL5 and PM5D, Avid VENUE Profile and SC48, and DiGiCo SD8 digital models.
“It was my first time on one of DiGiCo’s newer consoles,” he states. “I’ve mixed on the D5 so I’m familiar with the platform, but it’s a different control surface, so I downloaded the offline editor in advance, put together an I/O template and emailed it to the production manager to load onto the desk.”
He leans toward the Avid VENUE platform. “They provide an immense variety of plug-ins, and although I generally don’t use a lot of them, the ones I do use just aren’t available in other platforms unless you have a Waves bundle,” Frey says. “I really like the Universal Audio LA-2A for bass. Other than that and multi-layered drum compression, the effects I use for this group is very basic – just two reverbs, one for the drums and one for everything else.”
The latter is largely used for vocals, but Frey also employs it for solos “to keep everything in the same space, sonically.
“I also apply a delay for vocals and sax solos, and a pitch effect for vocal thickening, but it’s very subtle,” he continues. “Everybody sings, Alan, Alastair (Greene, the electric guitar player) and Todd (Cooper, the sax player) double on lead vocals for a few numbers, and with seven vocalists and seven-part harmonies, the vocals are big already.”
Front of house engineer Martin Frey (left) with house A1 Eric Hatcher at a Yamaha CL5 at a stop at the Sunrise Theater in Fort Pierce, FL. (click to enlarge)
Frey usually tours with his own nearfield monitoring package for front of house – a pair of EAW MicroWedge MW10s driven by an amplifier and an EAW UX3600 3-in/6-out digital processor. This time out, however, he kept it simple by using headphones, a matter of expediency.
Favoring The Overhead
The tour carried a straightforward microphone package, including models from Audio-Technica, which Parsons endorses. A-T ATM350s cardioid condensers were clipped to the drum toms, with AT4033 large-diaphragm condensers utilized with guitarist Greene’s Marshall cabinet.
“The ATM350s travel with us everywhere,” Frey details. “I like using condensers on rack and floor toms. They tend to have a flat response and bring out the actual sound of the drums and drummer, and our drummer, Danny Thompson, is very dynamic. We can have a 25 to 30 dB range in level on the drums, which means I have to be very careful with gating because I don’t want to miss a note.”
To fill out the drum package, Frey requests a Shure Beta 52 for kick and SM57 for snare based on their wide availability and consistency, as well as the fact that they can take substantial abuse and still perform well.
Parsons chose to be positioned on a riser behind the front line. Photo credit: Douglas Wells. (click to enlarge)
“I’ve never really felt the necessity to have more than one microphone on a snare drum,” Parsons states. “If there’s not enough top-end snap, I prefer to add EQ rather than mic the snare bottom. I think using EQ is more natural and, after all, when you’re listening to a snare in a room you don’t have your head underneath it, do you? If you want it to sound like drums, you’ve got to favor the overheads and tops of the drums.”
Similarly, he wasn’t convinced that two mics on the kick would make a “night and day” difference, given the ample low-end support offered by the subwoofers included with the systems encountered on the tour. Shure SM81s captured hi-hat, and Frey chose to carry a Shure VP88 stereo condenser for overhead to capture imaging, which is also helpful to monitor engineer Phil Dunn’s in-ear monitor mixes.
All vocalists were supplied with Shure SM58 microphones, with Cooper (again, the band’s saxophonist) given his SM58 wirelesslesly. Frey also used an SM58 for Parsons’ vocoder, patched through the unit’s XLR out. Most direct (DI) boxes for Manny Foccarazzo’s keyboards as well as numerous players’ acoustic guitars were supplied by the house or rented. But Frey carried a Radial JDI specifically for Guy Erez’s bass, and he also deployed a pair of Optogate PB-05 automatic mic gates on the vocoder and Erez’s vocal.
“The Optogates work well, and although we could probably make more use of them, if you open up an Optogate, it does change the sound,” Parsons notes. “Manually, you can ease the mic in. When the Optogate kicks in, you’re hearing drums, for instance, on that vocal mic, so it can be problematic. I think we’ve kept it under control, but if you had the whole band on Optogates, the sound would change constantly.”
Similar To Symphony
Frey’s preferred touring house system includes EAW Red Certified KF740 line arrays, which he endorses based on the performance of the loudspeakers and the tool kit that accompanies the system. This includes Resolution array design and modeling software, Greybox DSP settings with EAW Pilot in DSP-capable amplifiers housed in Powercube racks and use of Powertools for on-site system optimization.
While that package wasn’t feasible this time out, Frey says he was still able to achieve consistent sound relatively easily at each tour stop. It helps that the band members have substantial experience as both live performers and studio musicians. “Not only do they know a lot about sound, but they know a lot about their own sound,” he observes. “Again, my whole approach to mixing this show is much like that Alan took in the studio, and musically, this show has a huge dynamic range.
“For example, the song Sirius, which starts with huge drums and guitars, comes down to almost nothing, segues into Eye In The Sky, and then ramps up again to the chorus and those seven-part harmonies. It’s very similar to the dynamic range of a symphony performance.”
Monitor engineer Phil Dunn working pre-show at a Yamaha PM5D console. (click to enlarge)
Like Frey, monitor engineer Dunn encountered a variety of consoles on tour, but in the U.S. he had the luxury of using the same Yamaha PM5D at six shows. “That’s his preferred workflow environment,” Frey says, “but I’ve seen him mix on just about everything and we both have show files for virtually every desk.”
Dunn supplied seven in-ear mixes for the band via Sennheiser ew 300 IEM G3 systems. Parsons sported ACS T1 Live triple-driver ear monitors, with the band on a mixture of Future Sonics Atrio and Shure SE425 earphones (the latter for lead vocalist PJ Olsson and drummer Thompson). Dunn also developed mixes for wedges and side fill, kept at low volume, and a mix for two wedges and a sub for Thompson.
“A monitor engineer’s job is a performance, every bit as much as a front of house engineer’s job is a performance,” Parsons notes. “The quality of the playing and dynamics are, to a certain extent, affected by what we hear, so the monitor engineer is every bit as important as the house engineer to most bands. The mix we hear is an inspiration. If it isn’t right, it’s counterproductive.”
For a show at Ruth Eckerd Hall in Clearwater, FL, APLP was joined on stage by a 29-piece orchestra. “The band’s stage sound isn’t loud to begin with, which helps, but the challenge was bringing those additional mics into the mix in front of a pop/rock band,” Frey says.
The Midas H3000 and Yamaha PM5D that teamed up at front of house for the show that featured an orchestra. (click to enlarge)
The 16-piece string section was close miked with ATM350 condensers, with dynamic mics deployed for section miking of the brass and horns. Additional instruments included tympani, woodwind soloists, English horn and vibraphone/xylophone.
He mixed the orchestra on a Midas Heritage 3000 console, sending stereo and mono stems from it to a Yamaha PM5D. “I just turned up the gain on each input on the H3000, engaged the high-pass filter as required, and brought the faders up to achieve the correct balance and sub-mixed strings, horns, percussion and solo instruments into the PM5D,” Frey details. “Everything was flat, no EQ whatsoever. Again, that was a benefit of the flat response of the condenser microphones and their accuracy in picking up the acoustic sound of the instruments, which, really, is the same way you’d record a symphony.”
While there are obviously differences between mixing a live show and in a studio, Parsons concludes, there are commonalities as well. “At the end of the day it’s about balance, and the way you achieve a good balance is similar. You’re more likely to use noise gates live, if you’re constantly fighting separation issues, and with live there are feedback issues you don’t normally experience in the studio. But other than that it’s a channel per instrument, pushing faders up and down, and making it sound pleasing to the ear.”
Based in Toronto, Kevin Young is a freelance music and tech writer, professional musician and composer.
Updated Master Fader Control App For Mackie DL Series Mixers Now Available
Updates include choice of vintage or modern dynamics and improved snapshot operation
The latest update to the Master Fader control application for the Mackie DL1608 and new DL806 digital live sound mixers is now available for free download from the App Store here.
Master Fader v1.4 launches for iPad with a host of new features, including a new selection of channel plug-ins and improved snapshot operation. It also offers an entirely new selection of vintage processing. After analysis, development and testing, the new vintage EQ, compressor and gate deliver the musical characteristics of favorite industry-standard processors.
The vintage compressor and gate provide a lightning-fast attack, non-linear attack/release and program dependent release making them great for drums and other transient sources. The vintage EQ captures the unique interactions between the shelving bands and the gain and frequency-dependent Q structure of the parametric band to deliver the clarity and sheen of classic EQ designs. The EQ is good for getting vocals, guitars and other instruments to sit perfectly in a mix. Now, DL users can choose between clean, surgical modern processing and the new range of classic vintage plug-ins for more flexibility in use.
In addition, a huge update to the show management and overall snapshot operation makes DL mixers even better for users with complex shows and multiple wireless devices. Users can snap a photo of the band with their iPad camera or choose from a selection of handy icons and use this as channel ID. the images are now synched between all wireless devices connected to the mixer.
The snapshot system will also store and recall these images across devices and the user can add an image to each show to help with identification. Plus, shows now load in the background, keeping all wireless devices online when a show is changed. And shows will seamlessly transfer between the DL1608 and new DL806.
Master Fader v1.4 is now available as a free download from the App Store here.
Posted by Keith Clark on 04/08 at 05:01 AM
Friday, April 05, 2013
Roland Systems Group Releases iPad App And Software Update For M-300 V-Mixer
Roland Systems Group (RSG) is pleased to announce the immediate availability of a dedicated iPad Control App and V1.5 software update for their popular M-300 V-Mixer console.
Roland Systems Group (RSG) is pleased to announce the immediate availability of a dedicated iPad Control App and V1.5 software update for their popular M-300 V-Mixer console.
The M-300 V-Mixer console is a core component of the V-Mixing System providing mixing, effects and external control of digital snake pre-amps, multi-channel recording, instant playback, rehearsal and RSG’s Personal Mixing System for musicians.
Some of the new features in this free version update includes a new 31-band mono GEQ, an expanded group of library effects, cross-fade for scene changes and a detailed Recall Filter function that enables you to select which parameters to recall at a greater level of granularity.
The new software also supports the Roland Wireless LAN USB Adapter enabling wireless connectivity between the M-300 iPad Control App and the console.
The iPad application is designed to control parameters of the M-300 allowing the user to store scenes and edit the Channel Strip (EQ), GEQ, and Sends on Fader to name a few.
Using Wireless LAN to connect, users are able to adjust mixing parameters on stage or around the room, away from the M-300 positioned at the front of house.
By connecting Roland’s new Wireless Connect USB adapter to the M-300 USB port, the M-300 appears on a wireless network enabling the iPad to connect and control the console.
Other V1.5 operability enhancements include a CHANNEL DISPLAY screen for DCA groups, the ability to disable more user settings, and a default guest startup mode to allow basic user functionality without admin level control. For monitoring, a dimmer function has been added and a lock out feature to disable the LEVEL knobs to prevent accidental monitor volume changes.
The Roland M-300 Version 1.5 upgrade has improved the number of RS-232C control parameters ensuring that system installers, integrators, and users have access to more remote control functionality from touch panels, video devices, and software.
Concurrent with this update is the announcement that all V-Mixers now include the ability to do multi-channel recording. By simply connecting a Cat5e/6 cable from any REAC port to a network port on a PC enables 40 channels of WAV capture.
The M-300 Remote app is available from the iTunes App store on April 4, 2013 or can be downloaded here.
Roland Systems Group
Thursday, April 04, 2013
Church Sound: Mixing For The Whole Sanctuary
Sometimes the best way to achieve the most even coverage, and understanding of the mix, is to get out of the booth!
Ever had a comment from a worshipper, whether positive or negative, regarding the live sound experience – and it differs totally from what you thought you just heard and mixed?
Large room acoustics (particularly room modes), loudspeaker selection / orientation / optimization, audience size and participation, and several other factors all contribute to the fact that the live sound experience is different in every seat in your worship space.
If it is a great room with proper system design and installation, those variations may be minor. In many instances, they are not minor.
Either way, they do exist, and the FOH mixer must realize that he or she is only listening to (and mixing to) one position’s perspective when standing behind the mixing console.
During worship, only one of all those factors is under his control – the mix. The best the mixer can do is understand the other factors and learn to mix within that particular environment.
There are some worship facilities where consistency has been achieved across most of the audience area through excellent design and integration…but for the vast majority of venues, it’s one thing to create a brilliant mix for the mix position and another thing to translate that across the whole house.
So it is critical to walk the audience area whenever possible to hear the perspectives of the audience areas (especially if there is a trustworthy A2 to drive the console for a few minutes at a time). Tonality may be noticeably different in some locations.
For instance, it may be discovered that the majority of the house hears a little more bass thump than the mix position does. The mixer that notices this can take it into account in the mixing process. That would never be noticed, and compensated for, without walking away from the mix position.
In addition to tonal variations, it is not uncommon that loudness changes with position as well. If the loudest locations are in the front rows, that may be ok.
Wouldn’t even the least technical worshipper expect a bit of a louder experience when choosing a front row seat? Consider that the overall worship level should be mixed for the loudest location in the house. If that is not the mix position, then periodic walks are necessary to ensure excessive loudness does not occur at any seat (or the complaints that follow).
If the mixer can only walk the house during sound check or review, ok. If he can walk the house discreetly during the live service, even better. Not only does the presence of the audience acoustically affect the result, but an audience participating in corporate worship (singing) markedly affects the overall sonic experience.
For this author, nothing replaces the value of briefly walk-checking the house during the live worship mixing experience.
The varying parameters discussed above, within which we must operate, are mostly results of room design and system design or optimization missing their marks.
But rather than blame those factors, learn them, and mix around them.
If they are to be addressed and improved, that is for another time (and is off topic here).
So, next time you receive a comment regarding the sound experience in worship, whether positive or negative, make sure you ask where the person was sitting.
That can help greatly in understanding and interpreting various perspectives.
And for those mixing on a “stereo” or other multi-channel format PA, here are a few additional tips:
Imaging and Localization
The majority of PA systems in our nation’s churches are not stereo, including a number that are actually described as such. Some claiming to be of the stereo format are actually mono, dual mono, or something else.
Successfully implemented stereo and LCR (left-center-right) systems are wonderful and, while the definite minority, are slowly becoming more common in the worship community.
But whatever the format, if you mix worship on a PA system where the “pan-pot” control on your console does affect the positioning of the sources in some way, even pseudo stereo or some other multi-channel format, it becomes incredibly important to walk the entire house while considering imaging and localization.
For instance, what happens if you pan the floor tom “hard left”? Do all the seats hear it hard left from their perspectives, or from some other direction?
Do they hear it at all?!
Walking the house is no longer just for level, tone, and balance checks.
Good imaging in a mix can be a splendid enhancement for the worshipper. But mixing multi-channel sound reinforcement requires knowledge of how imaging is being conveyed at every seat in the house.
Localization should be carefully preserved (the brain naturally wants to hear and see a source from the same direction).
For instance, an interesting stereo effect created (and listened to) at the mixing booth may be impressive - but may also be a sonic disaster at other locations in the house! Know your system.
On a multi-channel PA system the mixer has some control over image and this comes with added responsibility (and by the way, multi-miking tricks take on a new role).
Just as with loudness and tonal variations, walking is the way to understanding how panning decisions are translated to the majority of listeners.
For more worship audio tips and techniques visit Sennheiser.com.
Wednesday, April 03, 2013
Church Sound: Communicating With The Worship Band
Do you have a tried and true method of communicating in the midst of a service?
With the rise of personal monitor mixers, the Sunday morning sound tech has had to worry less and less about the monitor mix for the musicians.
However, there are still a large number of churches which either do not have personal monitor mixers on stage. Or, only some musicians have personal mixers and some musicians like the vocalist still have their monitors controlled at front of house.
One very compelling reason to keep control of vocalist monitor mixes at the main mixing board is feedback.
With the potential for feedback, the front of house sound engineer/tech has a few options.
1. Control all monitors that have open mics near them.
2. Work with and educate the musicians on how to properly use a personal monitor mixer, and help them understand what causes feedback.
3. Put headphones on the musicians who have open mics in order to remove the potential feedback issue.
For the large number of churches who don’t have personal monitor systems, and for those (like me) who still control vocalist monitors from the front of house console, nothing can be more embarrassing and humiliating than having a musician ask for more in the middle of a worship service.
To help avoid this, I suggest working with the musicians to develop a simple set of signals to know when and how to adjust their monitors.
Giving credit where it is due, it was a worship leader who approached me years ago and explored setting up some signals. He had a great spirit about the situation, explaining that he understood that I couldn’t listen to monitors on stage to adjust level.
This worship leader added, however, that I didn’t know his mood or physical condition on any given morning, noting that when his allergies are bothering him, he not only needs more overall level, but just as important, he needs more piano to stay on pitch. He said that when he’s congested his own voice seems to “resonate” in his head, and when this happens, he can’t hear the piano as well.
So to help each other, we worked out signals.
If he held the mic and pointed his finger on the hand holding the mic up towards the top of the mic, it meant needed more voice. If he clutched the mic with two hands, then he needed more piano. Finally, if he put his hand by his side parallel with the ground, he needed less overall monitor.
These three simple signals served us very well the numerous times that we worked together.
Additionally, no one (unless they were very astute) in the congregation knew that this was taking place.
As a side note, a good number of years ago a vocal group sang at a church where I was involved with the tech. During the rehearsal time, I began to pick up on the somewhat complex set of signals they had with the sound tech.
A raised outstretched arm met one thing, a raised arm with a finger pointed in the air meant something else, and two raised arms meant something different.
I figured out that their movements not only controlled monitor levels, but also song list order, when the offering would be taken and they even included a request to have someone get more water bottles.
To the audience it was totally transparent. I even heard some say how “charismatic” the group was with their arms always seeming to be in motion. If only they knew…
From my simple example to the more complex one about the vocal group, a few important things made the signal system work.
1. Most importantly, the sound tech was constantly watching the stage. No head buried in the board action here, but rather, heads up and pay attention!
2. This seems obvious, but both parties need to know and understand the signals. I occasionally did get what I thought were conflicting signals, but usually responded with the correct action.
The conflicting signals happened with the worship leader accidentally made a signal and then quickly corrected it by making a countering signal that negated the previous one; like more vocal in the monitor followed by less vocal in the monitor.
I understand that particularly for the person on stage, it’s easy to accidentally give a signal or mistakenly give a wrong signal, so as sound techs, we must be gracious and respond to all requests. This will build trust.
3. As with everything related to worship sound, an understanding and gracious spirit from both sides of the stage is essential.
Doing so will ultimately help both of you out and will also bless the congregation because they won’t have to endure the interruption of a musician asking for his monitor to be adjusted.
Gary Zandstra is a professional AV systems integrator with Parkway Electric and has been involved with sound at his church for more than 30 years.
Tuesday, April 02, 2013
Midas Previews Generation 2.1 Digital Console Software
Adds a rack full of new latency-compensated FX plug-ins and dynamics processing options
Midas has announced the first public viewing of Generation 2.1 software for its entire range of digital consoles. It will be on display at the upcoming Prolight + Stage 2013 show in Frankfurt, Germany.
Midas Generation 2 software, unveiled last year at Prolight+Sound in 2012, added a host of new features and functionality to Midas digital consoles, including new FX and dynamics options, plus new mixing concepts such as Midas’ unique MCAs and “collapsed” fader flip.
Now Midas is pushing the boundaries once more at PL+S 2013, adding a rack full of new latency-compensated FX plug-ins and dynamics processing options to every console in the PRO series, plus the flagship XL8.
New FX plug-ins include a dual-channel Klark Teknik DN60 Real Time Analyser, a K-T tape saturation effect, sub-harmonic generator, multi-channel input phase adjustment insert, a fifth input compressor option, ducker mode for noise gates, plus a new “transient accent” gate option and a variable presence control for all input channel compressors.
G2.1 will be available as a free download for all Midas owners.
“There has never been a better time to invest in Midas digital,” says brand development manager Richard Ferriday. “G2.1 is further proof that buying into the Midas brand is not just buying the world’s best-sounding mixer-it is also investing in constantly evolving products which offer great ROI, an industry-leading development team committed to providing no-compromise solutions, and well over 40 years of brand heritage.
“When combined with our recent competitive pricing policy, Midas now offers an unbeatable value proposition.”
Line 6 StageSource & StageScape A Great Fit For Mobile Church
Plenty of capability and scalability for the church's numerous applications
Casa de Oracion de Fullerton, a Disciples of Christ church in Southern California, faced several challenges when it came to upgrading its sound reinforcement.
Because the congregation meets in a shared space, they needed a solution that would provide easy and quick setup, great sound quality and, above all else, an increase in clarity and coverage. According to church chairman Jose Vazquez, the congregation found a solution to these needs with a Line 6 StageScape M20d mixer and StageSource loudspeakers.
Three separate denominations share the same sanctuary in Fullerton, each providing its own sound system. Casa de Oracion offers a contemporary music presentation with drums, bass, electric and acoustic guitar, keyboards, and up to five vocalists. The combination of StageSource loudspeakers and StageScape mixer offers the right blend of portability, power and truly superior sound quality.
“First, we did a demo of the speakers. We tried several brands, but Line 6 sounded the best. We bought two of them, because they were such a big improvement compared to our old system,” says Vazquez. “We got lots of compliments. Then the people from Jim’s Music in Tustin, CA came out and demonstrated the M20d mixer. We liked everything about it, with the touchscreen control and easy, digital scene recall. It also had more channels than the little mixer we had before, plus it can be controlled from an iPad. That was important to us.”
The church invested in the StageScape M20d mixer and added two more loudspeakers: one L3t for use as a monitor, and one L3s subwoofer. When he set the system up for the first time, Vasquez grasped the total capabilities and scalability of the Line 6 system.
“The mixer knows what speakers are plugged into it, and how they are being used—subwoofer, for PA or for monitors,” he notes. “And the scene even tells you what channels to plug the instruments into. That is very convenient when you only have a half hour to set up. It also has automatic feedback control. But the best part is how easy it is to get a good mix.”
StageScape M20d offers a full-color touchscreen interface that enables users to quickly and easily increase qualities like brightness and clarity. “You hear the result instantly when you drag your finger on the screen, which makes it much easier for somebody who doesn’t use a mixer very much,” says Vazquez. “For me, learning to use the system was very fast, and having different scenes set up and ready for different services and events is a big help.”
The free iPad app and integrated multi-track recording functionality also appealed to Casa de Oracion. “Our mixing position is not in the sanctuary, so it is very hard to know what the mix sounds like in the audience. The StageScape iPad app lets me do all my adjustments from inside the room, with no need to run a snake through the room. That is a big improvement for us,” says Vazquez. “And to record our services is something we did not think we could get with our small budget. But all you need is an SD card, right inside the StageScape mixer. Push one button and you can record everything.”
According to Vazquez, the upgrade to the Line 6 StageScape M20d digital mixer and StageSource loudspeakers has been a total success. “For a growing church like ours, this is an excellent product. Everything is very fast to set up, very easy to use, and the sound is a big improvement over our old system. We are very pleased with the Line 6 sound system.”
Church Sound: Eight Tips For Improving Clarity In Speech
Optimizing the mix of the pastor's voice
There is a huge advantage of mixing music over mixing speech; you can blend sounds when mixing music.
That is to say, if you have one instrument or vocalist you can’t quite get right in the mix, you always have the other instruments and vocals to fill in and blend in with that particular problem channel. When it comes to mixing speech, i.e. the pastor’s voice, you don’t have that benefit.
Consider these eight tips for mixing the pastor’s voice and improving the clarity of their voice.
1. Consider volume and frequency. Vocal clarity comes from changes in volume and frequency manipulation. A pastor that’s hard to understand might only need a volume bump. Regarding frequency manipulation, clarity is found primarily in the upper mid-range frequencies.The typical frequency ranges used in the spoken word are; 150 Hz to 6,000 Hz for men and 350 Hz to 8,000 Hz for women.
2. Use a high pass filter (HPF) for dropping out sounds below 80 Hz. While a male’s voice *might* have frequencies in that low of an area, it’s nothing that’s going to help their clarity. Start with your HPF around 80 and slowly increase it up to the 125 Hz range. Wherever you find a noticeable change in clarity is the spot you need.
3. Boost in the mid-range. The important frequency range for speech intelligibility is in the 1,000 Hz to 4,000 Hz range. Often, a boost of 3 to 5 dB in this range will increase the clarity. Start around the 3,000 Hz point. If you have Q (bandwidth) control, use a wide bandwidth. In the cases where the 1,000 to 4,000 Hz range isn’t giving you the clarity you desire, consider going up to 6,000 Hz.
4. Add warmth to the vocal. A voice can sound clear but have no feeling behind it. This can happen with a voice that’s crystal clear but have little to no low-end. Add a 3 dB bump in the 160 Hz to 400 Hz range; lower for men, higher for women.
5. Remove sibilance. Sssssssibilance in vocals is when the sound of the letter “S” sounds more like a hissing snake. You can accentuate vowel sounds / add presence by increasing the EQ in the 4,500 Hz to 6,000 Hz range. However, the “S” sound lives between 5 kHz and 7 kHz. Therefore, be careful when adding presence because you can easily go from a great sound to a hissy sound. A de-esser can be used for dealing with sibilance, but I prefer first trying EQ changes.
6. Avoid distortion by adding compression. A pastor who is known for suddenly talking significantly louder is one that would benefit from compression. The problem with those outbursts is theycan cause distortion or simply the voice takes on different frequency characteristics as it gets louder. A compression ratio in the 2:1 and 3:1 range would be helpful. This way, their volume stays within a reasonable range.
7. Before doing anything, think about the pastor’s voice as that’s your foundation. I worked on EQ’ing a vocal that had a lot of low-end in the voice but also, surprisingly, had a noticeable amount of upper mid-range frequencies. Therefore, a wide mid-range boost made their voice sound worse. It wasn’t until applying a massive cut under the 350 Hz range, a narrow cut to a lower mid-range area, and a narrow boost to the upper mid-range that their vocals obtained the desired clarity.
8. Cut before boosting. A vocal will often have too much of something. Resolve issues with those “too much” areas before focusing on improving clarity via boosting.
Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians, and can even tell you the signs the sound guy is having a mental breakdown. To view the original article and to make comments, go here.