Here are some thoughts on doing virtual sound check if you don’t have a DiGiCo or Avid digital console at your disposal.
Disclaimer: This is not going to be exhaustive. There are hundreds of hardware/software combinations that will get you the same result. These are some ideas only.
Also, it should be noted that “cheap” is a relative term. All of these solutions are going to cost money, real money.
However, if you church is serious about raising the level of audio technician performance, it’s money well spent. On we go…
First, let’s define “virtual sound check.” It is simply the ability to record the band with each channel on it’s own track and then being able to play that recording back, in place through the same channels on your console.
To illustrate with a very primitive example, let’s say your “band” is a worship leader with an acoustic guitar. To facilitate virtual sound check, you would need a way to record the vocals and guitar on separate tracks, and you want those sources to come off the board before any EQ or dynamics.
Typically, you’re using direct outputs or the insert outputs. When you get ready to practice, you do a little patching (in software or hardware) and play back that recording through the same channels you use if the worship leader and his guitar were live in the room.
One thing should be immediately apparent here; the bigger your band (and the more sources you have), the more elaborate the system you’re going to need for virtual sound check. If you are running 30-40 inputs every weekend, this post is really not for you as that system is not going to be cheap.
Rather, I’m focusing on those who run fewer than 24 channels per weekend (a number that is not arbitrary, as you’ll see in a minute) and using an analog board. Here are a few ways to get it done.
Audio Interface(s)
The simplest way of doing this job is with a USB or more likely a FireWire interface such as the M-Audio ProFire 2626, a Focusrite Saffire Pro 40 or similar interface with 8 analog inputs and 8 analog outputs.
The first thing you’ll notice when shopping for an interface is that manufacturers get very creative in the way they count I/O. For example, the ProFire 2626 is listed as having 26 inputs and 26 outputs, which it does. But only 8 of them are analog.
M-Audio ProFire 2626
And if you’re using an analog console, that’s all you care about. If you have a digital console with ADAT I/O, you gain you an additional set of 8 useable channels.
Now, the catch here is that there aren’t any interfaces with more than 8 channels of analog I/O (at least I can’t find any). So that means if you’re running 12 channels of audio, 4 get left behind. Unless you get creative. You might ask why you can’t just connect two 8-channel interfaces to your computer and send those inputs to your recording software.
The issue is that most DAW software won’t support multiple I/O devices simultaneously. If your DAW of choice doesn’t support multiple I/O devices, there is a workaround, at least on the Mac.
In Audio/MIDI settings, you can create what’s called an Aggregate Device, which allows you to create a virtual device that is made up of two or more actual devices. You then chose the Aggregate Device as your I/O source in your DAW, and all the inputs and outputs on all devices that make up the Aggregate Device are available to the DAW.
So an example system might be made up of two Focusrite Saffire Pro 40 interfaces combined into an aggregate device and recorded using Reaper on a Mac Mini. That would give you 16 channels of recording and playback for around $1500, give or take. That seems pretty reasonable; at least until you consider the next option.
Focusrite Saffire Pro40
Hard Disk-Based Recorders
There exist on the market a couple of hard drive-based recorders, most notably the Alesis HD24. This little 3-rack-space wonder is capable of recording or playing back 24 tracks of 48 hHz, 24-bit audio.
The HD24 has 24 channels of analog I/O (plus 24 channels of ADAT I/O) and costs about $1600. Really, this is the way to go. It requires no computer, is simple to set up and operate and is rock-solid reliable. Add 24 channels of TRS patch cables and you’re done.
Alesis HD24
Other options include the Tascam X-48, which is a full-blown 24 channel workstation (and almost $5,000) and the excellent, but somewhat pricey JoeCo BlackBox, which will set you back almost $3,000 by the time you add a drive.
JoeCo BlackBox
Caveats
There are a few caveats with any of these solutions. First, if your board has direct outputs, it’s a fairly simple matter to patch those direct outs to the inputs of whatever recording solution you use.
Getting back in, however, will require some re-patching. You’ll want to pull your mic inputs, and patch the outputs from the recorder or interface(s) into the Line Inputs on your console.
If you don’t have direct outs, you’ll need to use the inserts. One cool thing about the JoeCo BlackBox is that the inputs are normaled back out to the outputs during every operation except playback.
That means that for recording (or just sitting there), the insert signal is returned and you can continue to use the board normally. When you hit “Play,” it opens the normal and sends the recorded signal back to the return on the board. From a user interface standpoint, that’s really nice. However, it will cost you twice what an HD24 costs…
When using the inserts, you will likely need to push the cables into the console until the first click. An insert jack is a TRS (tip, ring, sleeve) connector, so it has 3 contact points. Most consoles use the ring as the send, so if you push a TS cable in to the first click, you get the equivalent of a direct out (albeit an unbalanced one). Pushing it in all the way will interrupt the signal, so you’ll only do that on playback.
Using inserts is going to mean a fair amount of patching and some experimenting, so don’t decide to try this out at 8:50 on Sunday morning.
Once you get the system up and running like you want, start recording your services in all their multi-track glory. Then during the week, you can practice and experiment just like the band is there, only they aren’t.
Keep in mind, you won’t have any acoustic energy coming from the stage, so things like drums and vocals will be a little different. But this is still a great tool for training and experimenting with various processor settings.
Like I said, this isn’t exhaustive; I only intended to give a few examples. Hopefully though, it will get you thinking about how you can implement a virtual sound check system in your church.
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.
Simply put, if the various software and hardware products you’re using today aren’t going to be around in their current versions for the useful life of the sound recordings you’re creating (i.e. the copyright term), the archived recording must be prepared to weather that obsolescence.
The goal of preparing multitrack DAW data for archive is to minimize the layers of technology necessary to completely reconstruct the master recording in the future.
This article will introduce some basic techniques for creating both Consolidated and Flat Multitracks for archival purposes.
What Is A Consolidated Multitrack?
A Consolidated Multitrack is a digital audio fileset that completely expresses the EDL (Edit Decision List) information from a multitrack master recording. Specifically:
—Each DAW track is expressed as a single, continuous Broadcast Wave file (BWF); —All of the consolidated audio files share the same start times and durations; —All of the consolidated audio files share the same digital audio precisions, i.e. sample rate and bit depth; —All of the consolidated audio files share the same descriptive naming convention, e.g. trackname_songtitle_artistname.wav.
If all of the above specifications are met, a folder containing the consolidated audio files could be used to perfectly reconstruct the multitrack recording as far into the future as the Broadcast Wave file format remains viable.
Since the Broadcast Wave file is a widely accepted standard file format for media producers, its long-term viability (and eventual uniform migration) is virtually guaranteed.
Creating a Consolidated Multitrack:
1. From your last active session/project file, ‘Save As’ to create a discrete file from which you will create a Consolidated Multitrack.
2. Hide or delete any auxiliary signal path to simplify the working environment.
3. If additional Takes or Playlists are to be included in the Consolidated Multitrack, create new tracks to allow all of the source audio to be simultaneously visible/accessible.
4. Using session boundaries, location markers, or some other timeline tool, establish a repeatable global timeline selection that includes all audio from the earliest drop-in to beyond the longest running audio file.
5. Once your global selection is made, use the Consolidate or Merge functions to create a single continuous audio file that expresses the EDL information for each track.
6. Carefully, consistently label all of the newly consolidated audio files to reflect enough information that they could completely identify themselves by name, e.g. bassamp_take2_ohbabybaby_jimmysingsalot.wav
Once the above steps have been followed, a choice has to be made about how to present these consolidated audio files as a discrete multitrack recording for archive.
Minimally, a folder that follows the same naming convention as the consolidated audio files should be created to contain all of the associated audio files and metadata (like screen shots, rtf files containing session notes, credits, etc.). This method works fine, but will always require the multitrack to be reconstructed in a DAW for playback.
Alternately, a facility like Pro Tools’ ‘Save Session Copy’ could be used to create a new, independent playback session for only the archival material.
Using this method one would need to be careful to remove any non-archival audio and metadata from the source session before saving the copy.
This approach would facilitate more convenient short-term use of the archive, but doesn’t actually provide any additional content.
What Is A Flat Multitrack?
A Flat Multitrack is a digital audio fileset that completely expresses the EDL information from a multitrack master recording, but also expresses some subset of DAW metadata. What metadata is ‘flattened’ into the archive is up to you, your client, or contractual obligations, but it could include:
—Plug-in processing like amp simulation, ‘printed’ effects from auxiliary channels, or automated processing; —Automation data, like the fader rides on a lead vocal track; —Bounced submixes that would otherwise require reconstructing both complex routing and plugin processing.
It is critically important to note that a Flat Multitrack should never be archived instead of a Consolidated Multitrack, but only in addition. The Consolidated Multitrack is the master recording; the Flat Multitrack (when applicable) is an extension of that master.
Once a Consolidated Multitrack has been created, a Flat Multitrack can be created by repeating the process with a few additional steps:
1. From your last active session/project file, ‘Save As’ to create a discrete file from which you will create a Flat Multitrack.
2. Hide or delete all auxiliary signal path and metadata that is not going to be flattened.
3. If additional Takes or Playlists are to be included in the Flat Multitrack, create new tracks to allow all of the source audio to be simultaneously visible/accessible.
4. To flatten real-time processes like automation, time-based effects, or submixing, bounce/re-record the appropriate track outputs to new tracks, and remove the source tracks from the session. Note what metadata has been flattened.
5. Flatten additional metadata by processing audio files with offline versions of real-time plug-ins. Note what metadata has been flattened.
6. Make a global timeline selection, and use the Consolidate or Merge functions to create a single continuous audio file that expresses the EDL information for each track (including whatever metadata has been flattened into them).
7. Carefully, consistently label all of the newly consolidated audio files to reflect enough information that they could completely identify themselves by name, e.g. bassamp_take2_flatcompression_ohbabybaby_jimmysingsalot.wav
Since it would be unlikely that every track within a DAW project would have metadata worth flattening, there will likely be some tracks that remain in their consolidated form. I would caution that it would be both redundant and confusing to include these audio files in a Flat Multitrack archive.
Preferably, an additional folder of flattened audio files can be clearly labeled, and organized with the Consolidated Multitrack data. Future users can then reconstruct the Consolidated archive, and opt-in to any of the available, clearly labeled, flat content.
Contents Versus Carrier
It should be noted that this tutorial only addresses the form of the contents of a multitrack archive. The question of how to effectively store this information is an entirely additional- though related- matter.
Anybody who is serious about the subject should examine the Producer and Engineers Wings’ “Recommendation for Delivery of Recorded Music Projects” (pdf). It contains an example of a widely-adopted approach to redundant archival storage.
Rob Schlette is chief mastering engineer and owner of Anthem Mastering (anthemmastering.com) in St. Louis, MO, which provides trusted specialized mastering services to music clients across North America.
Engineer/Producer Matthew Noble Utilizes Metric Halo ChannelStrip On Recent Projects
For more than three decades, Matthew Noble has been at the forefront of pop music as a session guitarist, programmer, songwriter, engineer, and producer, with an engineering client list that includes Rihanna, Diana King, Southside Johnny, and Rod Stewart, among many.
These days, he performs most of his work out of the Loft Studios in Bronxville, NY and in the newly renovated Riverworks Recording in Dobbs Ferry, NY. Recent work with the musical Big River and gospel artist Rell Holland & Experience have put Noble’s new favorite plug-in, the Metric Halo ChannelStrip, through its paces.
“I tried Metric Halo’s ChannelStrip because some other people that I respect were using it,” explains Noble. “My friend Keith Brown, who is a well-known Nashville songwriter, was working on a project with Billie Decker, who is one of the hottest mix engineers in country music. Keith’s enthusiasm for the plug-in, together with his revelation that Billie uses it ‘all over the place,’ was enough to motivate me to check it out.”
Riverworks Recording boasts a huge, luscious acoustical space, which has changed the way both Noble and the producers and artists he works with approach the recording process.
“So much of my work there has involved tracking live instruments, as opposed to the ‘virtual players’ that live inside our modern computers,” he says. “While it’s been a refreshing change, it has also brought with it challenges. For example, getting a great drum sound and a great overall mix with the new expectations for how long things take these days is not easy.
“ChannelStrip has been very helpful because all the functions that I need to access quickly are all in one plug-in. These include the less ‘sexy’ functions, such as phase reverse and multiple trims, in addition to full-blown and flexible dynamics and equalization. Having everything in one plug-in has greatly improved my workflow.”
Noble often puts Metric Halo’s well-crafted presets to use: “The ChannelStrip presets are a great starting point. They’re especially useful in a time crunch, when the client is breathing down your neck. The acoustic guitar and drum presets are often spot on, right out of the gate. When I tweak, the informative GUI lets me know exactly what I’m doing.”
Of course, the best GUI in the world is useless if the algorithms behind it don’t cut the mustard. It’s here that Noble finds it really shines. “ChannelStrip has a great sound,” he said. “Like an SSL, it can be very aggressive and not at all subtle. Despite all its flexibility and sonic muscle, it has remarkably low CPU drain, which means I can use it whenever I need it.”
Lexicon has announced the availability of the individual plug-ins from its PCM Native Effects and PCM Native Reverb Bundles, with a total of 14 plug-ins available, including Pitch Shift, MultiVoice Pitch, Chorus, Resonant Chords, Random Delay, Dual Delay, Stringbox, Vintage Plate, Plate, Hall, Room, Random Hall, Concert Hall and Chamber.
“Offering the individual plug-ins from our PCM Native Effects and PCM Native Reverb Bundles represents our commitment to provide Lexicon users with greater flexibility and ease to obtain exactly the sound quality they are looking for from the specific plug-in(s) they need for any project,” says Rob Urry, vice president Harman Professional Division & GM of Signal Processing and Amplifier Business Units.
The PC- and Macintosh-compatible plug-ins are designed to work with popular DAWs like Pro Tools, Logic and Nuendo, as well as with any other VST, Audio Unit or RTAS-compatible host.
Each plug-in can be run in mono, stereo or mono in/stereo out, and on-screen input and output meters are provided for precise level setting.
All Lexicon plug-ins are Native only, and require iLok2 authorization. The individual plug-ins will be available in February 2012.
Unit Audio Announces Affordable New Line Of Passive Summing Mixers
Unit Audio has introduced the Milli-Unit and Micro-Unit, two new 8-input by 2-output compact passive summing mixers for studio/recording applications.
Both units are outfitted with eight balanced line-level inputs and two balanced microphone level outputs, all with Neutrik TRS connectors.
Input impedance is 20 Kohms, while output impedance is 220 ohms. Resistors are hand-selected, metered Xicon 1/4-watt, with 1 percent tolerance.
The units are hand-wired at the company’s headquarters in Nashville, TN, and are housed in rugged aluminum cases.
The Micro-Unit is also outfitted with two pan switches that allow for placing channels 1 and 2 in monaural (center), or hard left (channel 1), or hard right (channel 2).
“Is analog summing going to make your recordings sound like a Nashville studio with a billion dollars worth of equipment? Probably not, but you will notice a difference in your mixes using a Unit Audio summing mixer,” states Terry Auger, Unit Audio design engineer.
“Loosely quoting Shakespeare, one might say ‘To analog sum or not to analog sum?’” Auger continues. “This has been a point of controversy with digital recording for quite some time. With modern DAW software, mixing within the computer has resulted in some great sounding recordings, but I have long been intrigued by the concept of analog summing. I was not prepared to pay $800 or more to test that theory, so I engineered and built my own.
“Then to test the theory, I set out to see if there was any difference in the mixed sound. Much to my amazement and pleasure, I did notice a subtle but very pleasing difference in the stereo separation and placement of the instruments compared to my ‘in the box’ mixes.”
The Milli-Unit is priced at $149, while the Micro-Unit carries a price of $189. Both units can be ordered directly from the company website.
This article is about a shift in perspective mostly. It’s about really embracing the tools you already have and using them to their fullest potential.
For example, I have a 12-channel Mackie mixer that I double as an analog external summing amp. It contains FX sends and returns, allowing me to utilize any number of effects units (not to mention plug ins), which can be brought back in to the stereo bus or run parallel.
I also use this mixer to control every piece of audio/visual equipment in my apartment from my bedside.
Thus rises the essence of my point. If you look, there is most likely more at your fingertips than you would realize.
Here are a few other ideas:
Digital Distortion
Exceeding 0 dBFS is not necessarily always a bad thing. In fact, it can be used in many recurring situations. I often use it as an effect to gain an overload distortion quality which may sometimes work inherently with guitar tracks, certain vocal tracks, and definitely synthesizers.
It can also be used to achieve a distorted lo-fi effect or an over-compressed quality without using an actual compressor.
Elastic Audio
One of the main goals when using elastic audio is to mask the fact that you are using elastic audio. Instead of taking this approach, try embracing the imperfections inherent to the recalculation of waveforms. Stretching or compressing waveforms drastically can produce extremely unique processing errors, using a monophonic algorithm for a polyphonic instrument can produce interesting aliases, and using the varispeed algorithm on anything can produce seemingly infinite outcomes.
Buffer Size
For a unique glitch effect, drop your playback engine’s buffer size to its lowest playable setting and record the output to a separate track. This allows you to practically “audiosuite” a decrease in buffer size for specific instruments (or across the board if you’re feeling frisky).
Delay Compensation
Delay compensation can be used similarly to the buffer size. Additionally, when combined with precise time adjustments between different tracks, rhythmic feels can be solidified as well as totally phased out of control. There are many effects that can be attained, either in plug-in form, or using the “audiosuite” method.
A word of advice on opinions… f*ck everyone else.. try things for yourself.
Precautions
First of all, nothing is the end of the world (except the apocalypse) and there will always be more work (until the apocalypse). The music will go on (until the apocalypse).
That being said, there are very good reasons for a lot of the precautions taken within a DAW recording environment.
Now, I’m not necessarily saying go out and get the trashiest plug-ins that you can and try to figure out how to make them work for you (unless you want to, which would be kick-ass), but keeping this perspective allows you to work better with the tools you already have, and allows for your DAW to be a creative landscape, as a canvas on which to draw, rather than a harsh digital program to which you must adhere.
You and the DAW are just as flawed as each other in that your existence and the becoming of that existence are of imperfect origin. Therefore, just as you have to overcome imperfections in yourself, you must overcome the imperfections inherent to any piece of equipment. Besides, didn’t you get in this to push buttons, turn knobs and have total control? So take it.
Samuel O’Sullivan has been playing various instruments and composing within the bounds and mixtures of multiple genres for more than 10 years. He first established as a drummer/percussionist, has made his mark as a guitarist, vocalist, pianist, violinist, composer, and recording engineer. In addition to producing albums for various bands, O’Sullivan produces his own music under the name “A Mess of a Mind”.
At the NAMM 2012 Show, Behringer introduced the FIREPOWER USB/FireWire audio interfaces, the FCA610 and FCA1616.
The FCA610 and FCA1616 incorporate 24-bit/96 kHz A/D-D/A converters, support Windows XP/Vista/7 plus Mac OS X, and provide onboard phantom power for use with condenser microphones.
Due to its small size and low-latency operation, 6-input/10-output architecture, and two XENYX mic preamps, the FCA610 is a tool for traveling musicians who record and edit on their laptops. The portable FCA610 can receive power from a computer’s 6-pin FireWire bus or via the included external power supply.
Built-in MIDI I/O allows for easy connectivity with keyboards and other outboard MIDI hardware. All standard I/O formats are supported, including analog and S/PDIF (both coaxial and optical).
The half-rack-space FIREPOWER FCA610, which stows easily in a travel kit, can also be used as a premium 2-channel mic preamp and A/D-D/A converter.
With an expanded 16-channel I/O, four XENYX mic preamps and ADA8000 ADAT connectivity, the FCA1616 is more suitable for permanent applications as well as live performance multi-track recording rigs.
All standard I/O formats are supported; including analog, S/PDIF (coaxial and optical), ADAT and S/MUX, and a built-in MIDI I/O allows the user to connect keyboards and other outboard MIDI hardware.
The single rack-space FCA1616 also features eight analog Inserts for use of external effects such as compressors, gates and EQs. A dedicated power supply comes with the unit.
Included with both FIREPOWER interfaces is a massive software download at behringer.com that includes the widely popular Audacity audio editor, as well as a selection of audio software such as Podifier, Juice, Podnova and Golden Ear. Also included are more than 100 virtual instruments and 50 FX plug-ins.
FIREPOWER Features:
• Low-noise, high-headroom audio interface with 24-bit/96 kHz resolution
• Operates as multi-channel audio and MIDI interface via FireWire and USB2.0
• XENYX mic preamps with individual switches for Phantom Power, Pad, Low Cut and Hi-Z
• Direct Monitoring and Main Volume control on hardware front
• Two headphone outputs with individual volume control, mono and source signal select for flexible monitoring purposes
• Level control of stereo or 7.1 active loudspeaker systems with a single knob turn
• Smooth cross-fading between inputs and DAW playback signals
• Status and signal presence indication for all analog and digital I/O
• Standard port for Kensington security lock
A box of hand percussion instruments is one of the best investments you can make for your studio.
Shakers, rattles, tambourines and other clicky things can be added to just about any style of music from folk to electronic to heavy metal (hear White Zombie for proof of tambourine in metal).
You can use these instruments to fill out sparse arrangements, increase energy in a chorus, or add emphasis to certain beats.
Once you start building your collection it’s hard to stop, most small percussion instruments are inexpensive and many you can make yourself.
Having a variety of options will get you closer to the ideal sound for each song and minimal processing after recording.
As with any musical instrument, there is more to playing percussion than just shaking or smacking it.
Even the humble egg shaker is capable of a variety of distinct sounds just by changing hand position. I recommend watching some videos on YouTube for egg shaker, maraca, and tambourine for ideas and techniques.
One of the most important things in getting a natural sound from shakers, tambs, etc is distance. Ideally you record the performance in a large space with not a lot of acoustic treatment.
Hang the mic up high above the player pointed down. I have compared recording a shaker overdub in my control room versus the hallway outside with tile floor and it was dramatically different.
Having the reflections from the floor and walls helped create a more 3D sound even with one mic. Placing acoustic treatment behind the mic or on the sides around the mic sucks all the life out.
Any shaker tracks I’ve recorded close and in dryer environments (acoustically) have had a harsh, scratchy sound and were much harder to fit in a mix.
If you do prefer the sound of a close miked percussion performance play across the mic, rather than directly towards it for a more even low and mid frequency response.
Experiment with microphone options, condensers and dynamics will bring out drastically different qualities in percussion.
Condensers at a distance will capture a more realistic sound, FET models will pick up the fast transients more accurately than tube models. Dynamic mics react much more slowly and have a less accurate but still very usable sound.
Combining a few types of mics may help get you the perfect sound.
For music styles like indie rock, an audiophile quality recording of a tambourine isn’t going to be very helpful, you’re just going to have to distort and filter it later! Instead, experiment with different mics, tape recorders, toy mics and effect pedals to make things nasty. Besides being a lot of fun, it can be exactly what the song needs.
For processing these tracks I like short delays and reverb to create a doubling effect but it all depends on what the role of the percussion is in the arrangement, whether it should be drawing attention or just adding texture.
Close miked percussion tends to need more processing especially if you want it to sound natural (you see the contradiction there?). Using high and low cut filters to limit the spectrum to only whats necessary often helps when there are many of these parts.
Alright, now that you are prepared, make some noise!
Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com. To comment or ask questions about this article go here.
iZotope Introduces Mastering Essentials For Acoustica Mixcraft Pro Studio 6
iZotope has announced Mastering Essentials, a new tool designed specifically for Acoustica Mixcraft Pro Studio 6.
By licensing elements from Ozone, iZotope’s flagship mastering suite, Acoustica is delivering a basic mastering solution approachable for musicians of any level.
“Mastering Essentials is the perfect intro to mastering,” explains Alex Westner, iZotope director of business development. “With over 80 presets to choose from, users can get going quickly with great results. We take it a step further, though, by letting musicians experiment with customizing their sound.
“The three additional EQ, Reverb, and Tube Amplifier modules allow users to grow and expand their capabilities as they become more comfortable with mastering their own music.”
iZotope Mastering Essentials will make its debut as part of Acoustica’s Mixcraft Pro Studio 6, the latest edition of the company’s Windows-based DAW.
“Everyone trusts the iZotope name when it comes to mastering,” says Dan Goldstein, chief technology officer of Acoustica. “Mastering Essentials enhances our Pro Studio bundle with a top-notch tool for polishing any project.”
Mastering Essentials Key Features:
—Valve EQ module: high-quality 4-band EQ —Room Simulation (Reverb) module: controls for reverb and stereo widening, including a vectorscope for visualizing the stereo spread of the audio —Tube amplifier module: Bass Compression, a Tube Limiter, and Tube Saturation components —More than 80 presets, including presets for general use, special FX, audio enhancement, restoration, and over 40 different genre presets
Universal Audio Debuts Apollo Audio Interface With Realtime UAD Processing (Includes Video)
Universal Audio has introduced Apollo, a high-resolution computer audio interface that delivers the sound and feel of analog recording.
Apollo is a FireWire 800/Thunderbolt-ready interface combines genuine UA analog design with class-leading sound quality and onboard realtime UAD plug-in processing.
Designed to play a central role in a modern Mac- or PC-based studio, Apollo builds upon decades of UA’s analog hardware heritage. It offers extremely high-resolution sonics, with the lowest THD and highest dynamic range in its class.
Apollo’s premium mic preamps, top-end converters — and UA’s meticulous attention to circuit design — translate into greater accuracy and depth in recordings, from tracking and overdubbing, to mixing and mastering.
While Apollo’s “natural” sound is exceedingly open and transparent, it can quickly deliver a wide range of classic analog tones and color via its realtime UAD processing. Available with either DUO Core or QUAD Core processing onboard, this onboard DSP Acceleration allows for recording and mixing through UAD powered plug-Ins — with as low as sub-2ms latency — so that users can quickly monitor, audition, and “print” audio using classic analog emulations from Ampex, Lexicon, Manley, Neve, Roland, SSL, Studer, and more.
Apollo offers compatibility with Intel’s new Thunderbolt technology, as found on the newest iMacs, MacBook Pros, MacBook Airs, and next-generation PCs.
Available via a user-installable dual-port Thunderbolt I/O Option Card (sold separately), Thunderbolt provides lower latency, reduced audio buffer size, improved performance, and greater UAD plug-in instances versus FireWire.
And because Thunderbolt offers many times the bandwith of FireWire, it allows music producers to connect numerous devices in series with the Apollo interface — including hard drives, processors, and additional computer monitors — all with fast, flawless performance.
Apollo’s Core Audio and ASIO drivers ensure compatibility with all major DAWs, including Pro Tools, Logic Pro, Cubase, Live, and more.
Beyond this basic compatibility, Apollo’s included Console application and companion Console Recall plug-in (VST/AU/RTAS) provide control and recall of all interface and UAD plug-in settings within individual DAW session.
One key feature of Apollo isn’t really a “feature” at all. It’s the numerous design details that provide a fast, natural workflow. There are physical front-panel controls for all the most commonly used features, including preamp and monitor level knobs, channel selection, mic pad and low cut, phantom power, and even dual headphone outs with independent level control.
Smart Hi-Z inputs on the front panel detect when you’ve connected your guitar or bass, and automatically enable hardware and software monitoring.
Sonically, Apollo’s mic inputs and monitor outputs are digitally controlled analog, so you don’t lose audio resolution when you adjust gain. High-resolution/high-contrast metering, derived from UA’s legendary 2192 interface, is designed to be viewable at nearly any angle.
Finally, standalone operation means that you can use Apollo’s audio connections, and last-used DSP mixer settings, even without a computer connected.
Key features: —18 x 24 FireWire/Thunderbolt-ready audio interface for Mac and PC —Realtime UAD Processing for low-latency (sub-2ms) tracking and mixing w/ UAD Powered Plug-Ins —Premium mic preamps, top-end converters, and uncompromising analog design —Front-panel controls for all commonly used features —Full recall of interface and UAD plug-in settings within DAW sessions via Console Recall plug-in (VST/RTAS/AU) —8 analog inputs: 4 digitally controlled analog mic preamps; 8 balanced line inputs; 2 front-panel JFET DIs —14 analog outputs: 8 balanced line outs; 2 digitally-controlled analog monitor outs; 2 dedicated stereo headphone outs —10 channels of digital I/O: 8 channels of ADAT; 2 channels S/PDIF; Wordclock I/O —Dual FireWire 800 ports (standard) —Thunderbolt I/O Option Card for connectivity to new Macs and PCs (card sold separately) —Available in DUO Core and QUAD Core processor models —Includes “Analog Classics” plug-ins: LA-2A Classic Audio Leveler, 1176LN Limiting Amplifier, and Pultec EQP-1A Program Equalizer
Createdigitalmusic.com recently posted an article on a full-blown 48 track DAW for the iPad that looks pretty awesome.
The app comes from a new company called Auria, and isn’t available quite yet, but it’s features are very complete, as you can see from the following:
—48 mono/stereo, 24-bit/44.1kHz tracks, with recording for up to 24 tracks (you’ll obviously need a USB audio interface that can do that.)
—64-bit, double precision mix architecture (something that just came out in the latest version of Pro Tools)
—Full delay compensation
—“Vintage-inspired” channel strips, with a desktop-like UI and VU/RMS switching
—Plug-in support. Out of the gate, PSPaudioware, Overloud, Fab Filter and Drumagog all work. You need to do custom wrapping of plug-ins for this host; standard plug-ins won’t work. The format is based on VST, but it’s not VST in the traditional sense in that they have to be custom-wrapped for sale through the app.
—Dropbox, SoundCloud, AAF, MP3 export
—Advanced channel strips, EQ, expansion/compression and dynamic controls ready to go
—Convolution reverb. (This one blows my mind.)
—AAF import/export, making one definite application using this as a satellite for your desktop DAW”
Check out the video:
Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.
Producer Paul Rogers Adds SSL X-Desk And X-Rack To London Studio
Paul Rogers has installed a Solid State Logic X-Desk and X-Rack in his London-based studio. When in London, Paul works out of Pete Tong’s studio in the Matrix complex, Fulham.
Rogers regularly collaborates with Tong to remix artists as diverse as Gorillaz, Daft Punk & N.E.R.D., U2 and Bryan Ferry and on film scores including Michael Caine-led film Harry Brown. Rogers has also worked alongside the world’s best DJs, including Sasha, Digweed and Howells and has produced James Lavelle’s UNKLE Sounds live show.
“For years I’ve never really made the kind of records where a big desk is needed, he says. “I’ve been really happy with the sound I get for the music I make, and haven’t felt the need for a big desk in our world. But I had begun to think, now I’m doing bigger and more varied projects, it could be very useful. I think most people would love to own a big SSL desk but, apart from the obvious cost, where the hell would I put it? Then SSL launch X-Desk, which makes total sense for the sort of music I’m producing. I think with the shift in record sales these days, and the budgets slashed, the X-Desk gives you the best of both worlds, and the best thing about it is, it’s portable.”
“So we decided we were going to get some SSL gear in – an X-Rack that’s loaded with the infamous Bus Compressor alongside E-Series EQ and a Stereo EQ,” he continues.” And for summing and the like we use X-Desk. Everything’s patched, which is useful as we’re starting to get an addiction for analog synths.”
Rogers notes that the SSL gear has upped the ante in terms of sound. “The Bus Compressor, for example – it gives that ‘glue’ on a record that I’ve always missed. Without the Bus Compressor I always felt like you had to work that much harder to make things sit comfortably in a mix, but now it’s like having a magic button that just sorts your track out. We love the warmth of it, and also how you can just tickle it lightly or crank it if you really want to round off the edges.
“And we really love putting our analogue synths through those EQ modules. Putting the [Korg] MS20 through the E Series is a particular favorite. Just when you think you can’t get any more bass you put it through that EQ, dial up the bottom end and the pictures start falling off the walls.
“The SSL gear has changed the way I approach things in the studio. I think the main benefit that I’ve really noticed – apart from the nice, warm analogue SSL sound that I’ve always loved – is that the gear really helps you mix as you’re writing as it provides that natural gel in the record that makes all of the difference. Which, in turn, keeps things moving creatively as you really don’t have to work as hard to make things sound right.”
Keb Mo Named Keynote Speaker For 2012 AES Nashville Recording Workshop
The AES Nashville Section has announced that three-time GRAMMY-winning artist/songwriter/producer Keb Mo will be officially opening the 2012 Nashville Recording Workshop + Expo event on Friday, March 2, with a keynote address and interactive Q&A discussion with attendees.
Additionally, Mo and his engineer for both studio and live performance, John Schirmer, will be walking through the production of a complete song from start to finish in the final session of the first day of NRW+E 2012, giving their unique perspectives on both the technical and emotional experiences of making a recording and conveying the truest possible message on to the listener.
Having worked together on variety of musical projects, including production for artists on Keb Mo’s Yolabelle International label and musical scores for a multitude of television shows at his Stu Stu Studio in Nashville, this will afford a unique opportunity to gain insight to such a multi-faceted career and the creative process.
The Nashville Recording Workshop +Expo offers a compelling two-day recording workshop and gear exposition focused on getting the most from your personal studio recording environment.
AES Nashville, in conjunction with Audio Engineering Society, Inc. in New York is presenting this industry event on March 2 & 3, 2012 at the Rocketown event center at 601 4th Ave. South, near Nashville’s famed Music Row. Designed for recording musicians, songwriters laying down demos, and professional engineers working in a personal production space, the Nashville Recording Workshop +Expo will provide essential insight and information geared to boost your career and elevate creativity.
NRW+E presenters will include leading producers, engineers, acousticians, songwriters, and musicians sharing their professional techniques and knowledge in the areas of vocal and instrument miking, songwriters production of demos, arrangements and recording for better mixes, adding rhythm and spice with virtual tracks, collaboration across time and space, work environments that enhance creativity, practical acoustic and room treatment, and technical essentials.
Cosette Collier, chairman of the Nashville Section of AES. states, “We are thrilled to be able to offer the Nashville Recording Workshop + Expo once again this year. For two days, NRW+E brings musicians, songwriters, small studio owners, audio production students and home recording enthusiasts together with professional recording engineers and music industry professionals, not for the typical ‘How I Do It’ kind of sessions, but instead, ‘How YOU Can Do It.’ Inspiration is part of the experience, but “take away” information is key.”
Nashville Recording Workshop +Expo featured events and demos will appear in a live, on-stage setting on a regular schedule throughout both days, while exhibitor booths and displays complete AES Nashville’s own convention-type setting, featuring manufacturers representing all aspects of professional audio gear and services.
“Early Bird” registration for AES members and members of participating professional songwriter, performance, musician and engineering organizations is $79, non-members is $99, student members is $39, and non-member students is $59. Early registration is open through February 3, 2012. For the full program listings, information on registering or to book an exhibition space visit: www.nashvillerecordingworkshop.com.
Lately I’ve been listening to mixes from members over at MixWithUs.com.
A number of times I’ve suggested to people that they check their mixes in mono.
To clarify, everybody mixes in stereo. Stereo simply means the mix has two channels (left and right).
A mono mix is simply one channel. You combine (or sum) the left and right channels into a single channel.
Listening to mixes in mono can be very helpful. I’ll explain why.
Phase
Whenever you combine multiple signals together (especially similar signals), you run the risk of having phase issues. (See What is Phase?)
When using multiple microphones, whether on an acoustic guitar or a full drum kit, the more mics you use, the more careful you have to be.
As more mics are picking up the same signal, those signals, when combined, can cause cancellations at certain frequencies (if they aren’t perfectly in phase with each other).
When setting up microphones, it can be difficult to hear phase issues if you’re listening in stereo. Listening in mono lets you hear them more easily.
The same applies to mixing. You may have a really cool guitar sound panned to the left, and another great guitar sound panned to the right.
In stereo it sounds amazing, but in mono it suddenly becomes thin and hollow. It’s not necessarily a bad thing, but I’ve found that mixes that sound good in mono sound GREAT in stereo.
Bass Issues
This is especially true when dealing with the ever-problematic low end in your mix. You think you’ve done the perfect amount of EQ. The bottom end is full, but not muddy.
Then you check the mix in mono and BAM…it’s all muddy again.
Why? Because the tracks you have panned left and right in your mix all have little bits of low end that you don’t hear as clearly when spread out in stereo.
But when you “fold back” everything to mono, these little bits of low end add up to a big boomy sound.
While listening in mono, make the necessary EQ changes until everything sounds nice and balanced…THEN switch back to stereo. Wow! It’s pretty astonishing how good it sounds.
You didn’t know there were low frequency issues before. Listening in mono pointed it out, then when you fixed it and switched back to stereo, everything sounded cleaner and more professional.
How To Do It
There are lots of ways to listen in mono:
—Pan your master fader L and R sliders to the center. —Use a plug-in like TT Dynamic Range Meter on your master fader. It has a big “Mono” button. —Use a monitor management box, like the PreSonus Monitor Station. These have mono buttons on them as well.
Some people would tell me that mixing in mono serves no purpose, since people will always be listening to your mixes in stereo. Good point, but I view mixing in mono as one extra way to discover problems in my mixes.
Sometimes I’ll mix an entire song in mono (on accident), then switch it back to stereo right at the end…and WOW…it sounds amazing. Give it a shot sometime.
Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.
New Waves Audio Loudness Meter provides precision loudness measurement and metering for broadcast, movie trailers, games and packaged media.
The new Waves Audio WLM Loudness Meter plugin provides precision loudness measurement and metering for broadcast, movie trailers, games, packaged media and more. It is well-suited for content creators and post production houses as well as cable head-end facilities.
Fully compliant with all current ITU, EBU and ATSC specifications, the WLM offers comprehensive Momentary, Short Term, Long Term, and True Peak readouts, plus a unique warning and logging system that keeps track of levels, and lets the user know when they’ve been exceeded—or fallen short.
Features —ITU, EBU and ATSC compliant —Foreground, dialog, and average loudness measurements —Momentary, Short Term, Long Term, and True Peak readouts —Intelligent dialog sensor detects and measures speech —Mono, stereo, and 5.1 components
Measurement Methods —EBU uses foreground audio as the loudness anchor. —LM1 measures and averages loudness across the whole program. —DIAL uses dialog as the loudness anchor, measuring and averaging loudness only when dialog is detected.
Controls —Short Term displays (in LUFS) perceived short term loudness. —Long Term displays (in LUFS) perceived loudness averaged across the program signal. —Range displays (in LU) the overall loudness range across the program signal that —Momentary displays (in LUFS) momentary loudness levels, according to the Momentary scale control settings. —True Peak displays (in dBTP) inter-sample peaks which do not register in the sample data but may occur during reproduction of the digital signal. —Unders displays the number of times the signal goes below the specified Short Term Min value indicated in the settings panel. —Overs displays the number of times the signal exceeds the specified Short Term Max value indicated in the settings panel. —Measurements Play/Pause determines if program loudness is registered in the WLM memory for averaging long term loudness range. —Follow Transport determines if measurement starts and stops according to host application transport controls. —Timer indicates the amount of time measurements have been taken and integrated. —Reset resets the Long Term, Range, Overs and Unders counters and returns the integration timer to 00:00:00. —Method determines the measurement method. —Weighting determines the type of weighting filter. —Channel determines the channels being measured. —Short Term Max determines the maximum short term level. —Short Term Min determines the minimum short term level. —True Peak Max determines the maximum true peak level. —Target sets the target level reference. —Custom Pre Filtering provides low pass (LPF) and high pass (HPF) filters that pre-filter the audio prior to loudness measurement. —Momentary Scale determines the scale displayed in the momentary bar meter. —Logging displays CSV logging file options.
WLM is Native, SoundGrid compatible, available separately and in Waves Mercury, and requires iLok authorization. Mercury V8 owners covered by Waves Update Plan receive WLM at no additional charge.
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