I spent a weekend in the Chicago area installing an Apogee Symphony System for a client and doing some training. The install went surprisingly well, with no major issues aside from a missing BNC cable.
If you’re not familiar with the Symphony system, you should check it out. It’s a phenomenal PCI-based system that connects any Apogee converter directly into your DAW (in this case, Logic) with insane audio quality along with virtually no latency.
The system is a dream home studio setup, and it’s all centered around the Symphony system and a Toft ATB24 24-channel recording console (pictured above/left).
I have to admit, while I don’t use a recording console in my home studio, I grew very attached to the Toft board. It’s a great-sounding console with a ton of routing options.
There’s something about running an analog signal through an analog mixer that makes you feel like a real recording engineer. Of course, a great recording console requires a rack full of great converters and outboard gear.
In case you can’t quite make out the gear in the photo at right, here’s the list, from top to bottom:
click to enlarge
—Eventide H8000FW —Furman AR15 Voltage Regulator —Universal Audio LA610 —Two Empirical Labs Distressors —Apogee AD16X —Apogee DA16X —Switchcraft 64-point patchbay —Ebtech Line Level Shifter (8-channel) —A second Furman power conditioner
Pretty sexy, right?
So you might ask, “In a world full of such awesome outboard equipment and powerful recording software, why would anyone bother with a recording console?”
I’ve heard some big names in the recording industry say that Pro Tools can do everything a recording console can do, and that they would never use a console again. All they need is a mouse and keyboard.
With that in mind, should console manufacturers be worried? Is it really pointless to have a console in your studio? I think not. While you certainly can create a great record without a mixer, there was something about that Toft console that just sounded good.
In addition to the rack of gear, my client also had some nice keyboards — a Korg M3 with the Radius module, an Access Virus TI Polar, and a Roland V-Synth. We ran these directly into the console. All I can say is wow.
While these keyboards are all awesome, they sounded even more amazing through the Toft. There was this fullness and warmth. The more I pushed the fader up, sending the signal into the red, the better it sounded.
As you know, in any DAW, you can push the fader up, and the signal will get louder, but the tone won’t change, until that nasty clipping happens.
While a more educated author could tell you all the things that are happening on the console to cause this, what I can tell you is that I definitely noticed a “fuller” sound through the console. This will come in handy, especially during mixdown.
Rather than adjusting faders in Logic, my client will be able to run the signals out through the board, where there is significantly more headroom than in a DAW platform. It can give you a leg up on your mix, adding more punch.
Do we all need to go out and buy a console? Nah. But I have to say the Toft has significantly grabbed my attention.
Don’t count analog mixers out just yet. They’ll be around for a long, long time.
IK Multimedia Releases New British EQ Models In T-RackS Custom Shop
New EQ73 & EQ81 deliver sound of classic British consoles with added functionality
IK Multimedia has introduced two British EQ models in the T-RackS Custom Shop that provide the warmth and tone heard on numerous classic recordings, the new EQ73 and EQ81.
T-RackS Custom Shop is a Mac/PC audio plug-in and stand-alone mastering workstation providing a collection of accurate models of some of the most famous analog audio processors in music history, and it allows users to purchase what they need via online an à-la-carte purchasing system.
The new T-RackS EQ73 is based upon a classic British hardware unit. The original “Class A” device has a sonic signature almost instantly recognizable, with the signal passing through it becoming thick and bold, helping every instrument stand out with audible fullness. The T-RackS EQ73 model captures this nuance so that even in the DAW environment the tone stays intact.
The EQ73 is also designed to make sure the part of the preamp circuitry functionality was captured as well. The interaction between the preamp and the EQ section is what makes this EQ so desirable.
The wide gain range provides numerous “color shades,” and at lower settings it adds some harmonic coloration that will make kicks or snare fatter and punchier. Going up the gain hill moves it into downright distortion with an analog saturation rare for DAW plug-ins. It’s versatile enough to enhance any track, especially kicks, snares, overheads, bass, guitars and vocals.
The new T-RackS EQ81 takes the versatility even further; the EQ section is composed of four independent frequency bands, rounded out by high- and low-pass filters to eliminate unwanted frequencies from the signal. There’s also the ability of the high and low shelving bands to become peaking type if needed.
Both the EQ73 and the EQ81 models have the dual concentric knobs found on the original units “separated” for easier and faster shaping, and the unit offers improved readability of the knob positions for quick adjustment.
T-RackS Custom Shop allows users to demo any T-RackS processor for 14 days, allowing them time to evaluate the workings and sonic characteristics of the unit. When ready to purchase, users click the “Custom Shop” button in the bottom right corner of the processor to launch the T-RackS Custom Shop purchase system and complete the process in just a few clicks.
Sepura Plc Streamlines Radio Tech Development & Testing With Soundcraft Si Expression Console
Provides an assist in testing against specifications in typical usage scenarios, including background noise testing using surround sound
Sepura Plc, a designer and manufacturer of digital mobile radio products for public safety, industrial, commercial and military applications, recently implemented a HarmanSoundcraft Si Expression 1 digital console at its Cambridge, UK faclities to improve and streamline processes during product testing stages.
The Si Expression 1 is situated next to an acoustic testing chamber with a head and torso simulator, to test against specifications in typical usage scenarios, including background noise testing using surround sound. This system is set up to accommodate a binary goal: the automatic verification of products during late development stages and the manual acoustic testing during early development stages of software, terminals and accessories.
“These two activities have conflicting requirements and this is why we purchased the Si Expression,” says Edmund Elsey, senior acoustic engineer at Sepura Plc. “Our automated tests are run with in-house software that control our terminals remotely, which requires the signals to be routed to and from different equipment, depending on the test. We used a switching unit to route the signals, which not only introduced noise and ground loops, but also prevented easy monitoring. For manual testing, we needed to partially dismantle and reconfigure the system, as well as adjust signal levels and EQ.”
“However, the Si Expression allows for automated routing, monitoring of any input, and GEQ functions all in one,” he continues. “I now have all the equipment permanently connected because I can control routing from any input to any output using MIDI control messages sent directly from our software. For manual measurements, I can route any signal anywhere I like.”
The Soundcraft Si Expression is a flexible digital console that provides a value at its price range, with intuitive features such asFaderGlow, which illuminates the faders with color coded LED lighting. Workflow is taken to new heights, as monitoring, functions and effects are designed to be efficient and hassle-free. For example, Elsey found that he could use the GEQ to implement correction functions at any time through the bypass feature and customize the fader layout.
“I worked with a Soundcraft analog console before, and tried a demo of the Si Expression at Harman’s UK distributor Sound Technology, so I know that Soundcraft lives up to its name by manufacturing high-quality mixing desks,” he explains. “We also made sure that our high-powered digital radios would not interfere with the desk, as to not introduce TDMA noise onto the signals. In fact, I was impressed that Soundcraft sent me a copy of their EMC report when I enquired about RF immunity.”
“The Si Expression massively improved productivity in our test environment, because now the whole system can be configured at the touch of a button for different use cases,” Elsey concludes. “The THD and noise floor are also great and have been verified to be more than adequate for our needs.”
Ohio-based company has served as a Shure sales representative since 1939
The first meeting between McFadden Sales and Shure occurred in 1939, when founder Bill McFadden met S.N. Shure and decided to strike a partnership. This year, Shure is celebrating 75 years with McFadden Sales as a Shure sales representative.
“When you look back at the history of our company, there are a number of factors that have contributed to our success,” says Mark Humrichouser, general manager for the Shure Americas Business Unit. “One of those success factors is the strong relationship we have with all of our distributors and sales rep firms, like McFadden Sales. We greatly appreciate their efforts to not only sell our products, but also to represent and champion the Shure brand.”
McFadden Sales has been a premier sales and marketing company since 1938, serving the Midwest in representing professional audio equipment and musical instrument manufacturers. The company’s longstanding mission is to provide, maintain, and sell the best available products, with an emphasis on building long-term, profitable business relationships, reinforced by honesty and ethical business practices.
Based in Westerville, OH, McFadden Sales is headed by president Gary Dunaway, with Jay Dill serving as vice president, Carrie Walker providing senior sales support and Steven Sutherland serving as director of digital media. Territory managers include Dave Ray, Scott VanEaton, Mike Love, Mike Somerville, Andrew Yost, Andy Kerr, and Jeff Allen.
Q: My studio is outfitted with a DAW as well as a fairly large amount of outboard equipment, and it always delivers fairly good results. However, every once in a while there’s some weird distortion, and I’ll see a clip or something on one of the 20 different meters that’s either in the DAW or on the outboard. Is there some kind of hierarchy to these meters?
A: A situation where you’re using multiple pieces of outboard gear (preamps, compressors, etc.) in addition to a DAW can definitely lead to some confusion, as there are often numerous meters displaying very different results.
So if your preamp has a meter and your DAW has a meter, which one should you look at as a reference? Well, theoretically, you should keep an eye on both meters, but pay extra attention to the outboard gear meters.
Pushing a preamp or compressor too hard will evidence itself on the meter on that piece of gear, but may still look fine on the input meter of the DAW. And it can cause the recording to be distorted or over-compressed, or both, but the input meter of the DAW will still be bouncing happily at -3 dB.
The pitfall of not accurately monitoring the outboard gear’s monitors is that distorted vocals or squashed tracks can’t be undone. With gear that has multiple monitoring modes, like a compressor that can switch between gain reduction and output, it’s a good idea to keep an eye on both functions to get the truest picture of what’s going to the input of your DAW.
The real answer is to watch them all; however, make sure you’re watching not only with your eyes, but with your ears as well.
EQ is pretty simple, right? Crank a knob, hear the sound’s tone change?
Not quite. Just when you think you know everything there is to know about EQ, something new comes up. Here are a few advanced EQ techniques that you might not be using to full potential.
1. Mid/side EQ
Any true stereo sound might be able to be enhanced with mid/side EQ. It basically turns your stereo EQ into a frequency-specific stereo width adjustment tool.
You’ll get the most natural results by only processing the side channel. You can boost the top to increase clarity and dimension. You can narrow the mids to provide focus and punch. You can high-pass the bass to easily collapse the bass to mono without touching the mids and highs.
2. Spectral matching EQ
This one’s usually a multi-step process. First get the EQ to “listen” to some reference audio (such as another track or commercial mixdown), then get the EQ to “listen” to the audio you want to process.
Finally, the EQ can then either match the two (so that the processed audio sounds similar to the reference audio), or it can compliment the two (so that the processed audio sounds very different to the reference audio).
Matching EQ can be useful whenever you want one track to sound like another. Obviously, this might be useful in mastering, but it can also come in handy when working with samples from a variety of different sources.
Compliment EQ can be useful if you want to make sure two tracks do not interfere with each other.
3. Dynamic EQ
This is an interesting one. It allows the gain of each EQ band to change dynamically with the level of the audio. It can work a lot like a multi-band compressor, except that the envelope follower controls the gain of an EQ band instead of a frequency range. This allows you to get much more surgical and specific with how the audio is processed.
The most common use of dynamic EQ is de-essing vocals, where the high frequencies are turned down when there’s too much sibilance. It’s also useful for other situations where a recorded track needs to be cleaned up in a specific way, but static EQ or broadband compression are too blunt for the job. Things like low frequency bumps or thuds, or the occasional odd midrange resonance are sometimes good opportunities to use dynamic EQ.
4. Parallel processing with EQ
This is a good example of a more advanced pairing of EQ and compression. This technique works best with a naturally dynamic recording — such as a lead vocal or acoustic guitar.
Duplicate the track and heavily compress one copy while leaving the other relatively dynamic. Balance the two so that the compressed track is dominant during quiet passages and the dynamic track is dominant during the loud passages. This opens up a lot of interesting possibilities when the two tracks have different EQ applied to them.
For example, you could make the dynamic track brighter and the compressed track darker. It will sound as if the recorded instrument itself gets brighter in the mix as it gets louder. Or make the compressed track warm and full-bodied, but reduce the lower frequency energy in the dynamic track.
As the track gets louder, it thins out to make room for other instruments in the mix (which might also be getting louder), but stays warm and full during quieter parts where it’s more exposed.
5. EQ’ing effects returns
This is a good one, especially useful for reverbs. If you’re mixing in software, insert your favorite EQ after your reverb. If you’re mixing in hardware, bring your reverb back on a regular channel pair (not the less-featured stereo returns).
Or patch a decent outboard EQ after the reverb before it comes back to the desk.
Many reverbs have some in-board tone control, but it probably won’t be as flexible as your desk EQ or outboard EQ. This gives you much more power to shape the sound of your reverb and ambience at the back of the mix.
Happy with the reverb but it’s fighting a bit too much with the vocal? Give it a dip in the midrange. Want more focus in the low end of the slap bass while still retaining the ambience and space in the top? Bring in a gentle low shelf for a more natural cleanup than a low cut filter.
Mix sounding a bit dead? Add some more dimension by gently boosting the top without upsetting the mix balance.
And this is just scratching the surface.
All in all, there’s a lot you can do with EQ. Much more than might be obvious at first. Give these techniques a try and you might just find a new secret weapon that’ll save your next mix.
[Editor’s note: Of course if you want to rapidly improve your EQ skills, download Quiztones.]
Kim Lajoie is a Melbourne music producer specialising in composition, project management and digital audio technology. With over two decades of music behind him, Kim brings expert skill and wide-ranging influences to every project. Kim’s highly structured and disciplined approach allows his artists to advance their careers with confidence and determination. Kim takes care of the details so his artists can focus on their creative expression.
Also be sure to visit The Pro Audio Files for more great recording content. To comment or ask questions about this article, go here.
Love it or hate it, data compression (not to be confused with audio signal compression, which is a different issue) schemes like MP3, FLAC, AAC, and other relatives have fundamentally changed music as we know it. The battle between fidelity and portability was long ago declared no contest, with convenience winning hands-down over sonic quality.
But while the ability to compress what was once a wall full of vinyl into your pocket is a boon for music consumers, it’s a bit more of an issue for those of us on the creative side of the equation. As record stores gasp for breath, and online CD sales continue to decline, artists have little choice but to make their works accessible online, and that means crunching your music.
So how to best compress without making a mess of the music you’ve worked so hard to make sound good? As with most things, the answer depends on what you’re after, and what you’re willing to compromise.
When It Comes Down To Crunch
With the Internet as our primary resource for both selling and marketing our wares, the question is no longer whether to compress, but rather how, and how much. There’s a mind-boggling alphabet soup of compression codecs available, ranging from good to barely tolerable.
Far and away the most ubiquitous compression format is MP3 (which stands for MPEG Audio Layer III), and making your recordings available as MP3s is pretty much a given if you want maximum exposure. But not all MP3s are created equal; depending on the bitrate and other factors, an MP3 can sound nearly indiscernible from the original WAV file, or as flat and lifeless as a wet newspaper. Not surprisingly, there’s a direct trade-off between file size and fidelity.
The original iPod: the portable player that launched a revolution.
Briefly, here’s how MP3 (and most other compression schemes) work. The process employs a combination of digital technology and the science of aural perception (psychoacoustics) to remove data bits from the original digital file that are considered to be essentially inaudible. These bits can include frequencies beyond the normal threshold of human hearing, sounds that are masked by other sounds, and various other “redundant” sonic information.
The point of contention with this whole concept is just how much of that data is truly inaudible. While some bits can be removed with little consequence, much of what gets stripped away can subtly affect our perception of how things sound. While moderately compressed files can deliver near-CD quality sound, too much compression can remove elusive qualities that can make a difference to how we perceive music on a subconscious level.
What Have You Got To Lose?
With any compression, some audio quality loss is inevitable. Very high frequencies are typically the first data to be eliminated, and while in theory these sounds are inaudible, their loss can rob your music of its subtle overtones, presence, dynamic range and depth of field.
The audio resolution and sonic quality of an MP3 is determined by the bitrate at which it’s encoded. The higher the bitrate, the more data per second of music. As you’d expect, a higher bitrate creates better quality audio, along with a larger file.
Generally speaking, 128 kbps (kilobits per second) is considered the bitrate at which an MP3 begins to exhibit artifacts of data compression. Not coincidently, it’s also the rate many websites use for downloads, since it offers a smaller file size with relatively minimal loss. Rates below 128 kbps are usually not recommended for anything other than spoken word recordings. Bitrates of 192 kbps, 256 kbps, or higher preserve most of the original sonic information, making them a better bet for music you care about.
Another alternative is to encode using a VBR, or variable bit rate. VBR examines the data as it’s encoding, using a lower rate for simple passages and a higher rate for more complex ones. While the resulting file size is smaller than using a higher bitrate, sometimes VBR encoding can end up compromising the audio fidelity of delicate material like a solo acoustic guitar or vocal.
While MP3 is the most popular data compression format, there are countless other formats available. Each uses a different type of algorithm to determine what data to discard, and the resulting differences in sound can range from subtle to fairly obvious. There are far too many different encoding formats to name them all, but here are a few of the more commonly used ones.
WMA (Windows Media Audio) was created by Microsoft, and is often offered as an alternative to MP3 on music and video download sites. It’s also common on sites that offer streaming audio and video files compatible with Windows Media Player. While many people feel the sonic quality is superior to MP3, WMA files tend to sound overly bright and brittle, with less than optimal stereo imaging.
AAC (Advanced Audio Coding) was designed to be a successor to MP3, and although it is a sonic improvement, its popularity has never really taken off. AAC is a default standard for iTunes, the iPod, the iPhone, as well as PlayStation and Nintendo DS. It’s also often used as the audio component for Apple’s QuickTime and MP4 video formats. Generally speaking, if you’re going to offer a second encoding format online, AAC is the one to consider.
AC3 is a format developed by Dolby and is often used for video soundtracks due to its superior stereo imaging and ability to handle multitrack formats like 5.1 surround. Because of this, many consumer-grade DVD players support AC3 format.
RA (Real Audio) is a fairly good-sounding codec, but is on the decline due to the fact that the files only play on Real Audio’s proprietary player. Many people refuse to install the Real Audio player due to its excessive advertising and high-CPU demands.
Ogg Vorbis is an open standard audio format that delivers a very high-quality sound. Unfortunately, like many open standard projects, it has had a hard time catching on among users.
FLAC (Free Lossless Audio Codec) is one of the few audio formats that delivers truly lossless compression. FLAC is similar to a ZIP file, but is designed specifically for audio. FLAC is also open-source, and FLAC files can be played back on most MP3-compatible players. (There are several other lossless formats, including Apple Lossless, WMA 9 Lossless, Monkey’s Audio and MPEG-4ALS, but none offer the open compatibility of FLAC.)
To Squash Or Not To Squash
Many musicians and music aficionados bemoan the widespread acceptance of compressed audio, and rightfully so. Some recording professionals have begun to adopt the stance that, since their final product will be heard as an MP3 on low-priced earbuds, there’s little reason to bother striving for sonic excellence.
Thankfully, that attitude is still in the minority, and most artists still care deeply about the music they’re making. And for those artists, the good news is that as hard disk space continues to drop in price and broadband Internet access becomes faster and more commonly available, file sizes become less of a barrier. In fact, many musicians are now offering full WAV versions of their songs alongside MP3s.
In the end, what compression schemes you use, or if you use any at all, is strictly up to your ears and your bandwidth. The best advice is to listen very critically to whatever formats you opt for, and to select the highest quality you can. You can always compress larger files into smaller ones, but once you’re crunched those files down, there’s no way to get those bits back.
Daniel Keller is a musician, engineer and producer. Since 2002 he has been president and CEO of Get It In Writing, a public relations and marketing firm focused on audio and multimedia professionals and their toys. Despite being immersed in professional audio his entire adult life, he still refuses to grow up. This article is courtesy of Universal Audio.
The Radial JR2-DT is a dual-function control switch that enables the user to A/B toggle devices from the comfort of a desk-top position or remotely mute the system.
The new Radial Engineering JR2-DT remote is now shipping. The JR2-DT is a dual-function control switch that enables the user to A/B toggle devices, such as the Radial SW4 and SW8, as well as to remotely mute a system, from a desktop position.
“Since we launched the Radial SW8 backing track switcher, we have had several requests to produce a custom remote control to enable the user to reach over to switch from one backing track to the other should one of the playback systems fail,” explains Radial Custom shop manager Ryan Juchnowski. “We figured that since we were building an updated version of the SW8, it was the perfect time to produce an off-the-shelf solution. The JR2-DT is super compact, uses a standard XLR as the cable interface and is plug and play easy to use.”
The compact JR2-DT offers a choice of XLR and 1/4-inch remote outputs to suit various set-ups. There are two switches on the top panel: one switch is labelled A or B, the other mute. Each switch is a latching type that stays engaged until it is depressed a second time.
On-board LEDs illuminate to provide status. These derive power from the Radial device that is connected, eliminating the need for batteries or local power supply. The enclosure is constructed of 16-gauge steel with powder coated finish. As with all Radial products, the JR2-DT is made in Canada and comes with a 3-year transferable warranty.
True Colurs Mixes Bigger Projects With THE BOX From API
True Colours Studio adds THE BOX from API.
Located in one of the oldest towns in northern Italy, True Colours Studio is dubbed a ‘well-heeled project’ by owner Mauro Santinello.
What began in 2005 as a single recording and control room has now grown into a facility with three recording rooms for commercial audio projects, jingles, and most recently some major Italian recording artists.
To accommodate the needs of its growing presence in the music scene, True Colours has added THE BOX to its studio, citing its “high quality manufacturing, and the unique API sound”.
Santinello wanted a project console that would provide a cost-effective, versatile solution to handle the needs of his growing studio in the decades to come. Along with his booming presence in the recording artist scene, True Colours has a growing demand for complete post-production, musical arrangements, and the creation of music for movie productions and short films.
“I chose THE BOX because it represents the right console for a modern studio,” he explains.
The small-format recording and mixing console offers features that are not provided by most DAWs, including mic preamps, input signal processing, a high-quality mix bus, cue sends with talkback, and monitor control – all in a compact and versatile package.
True Colours has a growing number of external preamps, EQs, and compressors, which includes some existing gear from API. The ability to personalize setups was a major factor that drew Santinello to THE BOX. “I didn’t want a big console with 32 preamps and 32 EQs that are all similar.”
True Colours mixes and records music of all genres, with only metal and electronic music yet to make the list. The crew has recorded a wide-range of Italian artists including famed singer Zucchero, piano master Stefano Bollani, and jazz trumpet player Enrico Rava.
Most recently, Santinello used THE BOX for a session with singer Alberto Micaglio, whose acoustic presence is expanding from Italy, to London and New York. He has an album due out later this year.
“I have always wanted to buy an API console,” admits Mauro. “THE BOX offers a high-quality analog sound, and no compromises.”
Lectrosonics Digital Hybrid Wireless microphone technology is a key component to capturing sound for Pedal to the Medal.
George - Aris Anastassopoulos, one of the location sound engineers for Back2Back Productions’ Pedal to the Metal, is tasked with capturing the roar of the engine as the car passes the camera—challenging work with little margin for error. In order to accomplish it he relies on Digital Hybrid Wireless microphone technology from Lectrosonics.
Anastassopoulos has been working in professional audio for better than fifteen years—starting as a recording engineer for classical and orchestral recordings in addition to handling live sound reinforcement for open air festivals as well as surround sound system installations.
His location sound work has taken him to Germany, Spain, Italy, Brussels and the Greek islands for fourteen feature films, commercials, TV shows, and documentaries.
“Pedal to the Metal showcases host Brian Johnson’s expertise on specialty cars and driving and also gives viewers a glimpse at his personal life when he’s not performing on stage,” Anastassopoulos explained. “There’s a wide range of shooting scenarios. For this project, I’ve been using Lectrosonics SMB single battery and SMDB dual battery transmitters as well as an HM plug-on transmitter.
“On the receiving end, I use two SRb dual channel slot mount ENG receivers outfitted with the Lectrosonics Quadpack power and audio interface. I also use the Lectro RM app for Apple’s iOS-based smartphones and tablets, which lets me adjust transmitter level, change frequencies, or just put the unit into sleep mode.”
“I’ve been working with Lectrosonics gear for about one year and I’ve become very fond of the equipment,” Anastassopoulos continued. “The sound quality is excellent and nothing beats the rugged build quality.
“The transmitters’ Safety Combo Menu Locks offer protection from accidental menu selection and their backlit LCDs are a godsend during night shoots and in other low light conditions. I also love the fact that the HM plug-on transmitter is a great tool for wireless boom operation and for 48V phantom feeds.”
Anastassopoulos notes that Lectrosonics’ RF Scan and precise RF tuning steps add a great deal to RF agility and make the system very flexible and versatile in crowded RF environments. These features make the equipment very well suited for use in the UK.
“The UK has the world’s most crammed PMSE RF block (Block 606),” he adds. “Further, being able to use the receivers in true diversity mode or as dual single receivers is extremely useful. For one specific shoot, there was a lot of on-the-fly improvisations and a lot of unexpected performances—requiring on the fly decision making and flexibility from the crew.
“Being able to accurately monitor the receivers’ antenna reception strength provided considerable peace of mind and enabled strategic placement of my receivers for the best possible reception.”
Lectrosonics build quality gives Anastassopoulos peace of mind for what is unquestionably one of the most important chains of the location sound engineer’s setup—radio systems.
“Their constant upgrades and tweaks on the SRb means the company is constantly fine-tuning their product, improving the unit’s performance and value, and at the same time making it even more future proof,” he concludes. “I’m very happy that I made the switch to Lectrosonics.”
Three new plug-ins for UAD Powered platform and Apollo interfaces are part of new UAD Software v7.6
Universal Audio Direct Developer partner Brainworx has released three new plug-ins for the UAD Powered Plug-Ins platform and Apollo audio interfaces.
Available for purchase via UA’s Online Store for $199 each, the Chandler Limited GAV19T Amplifier, bx_refinement, and bx_saturator V2 plug-ins are part of new UAD Software v7.6.
Available exclusively for UAD-2 DSP Accelerators and Apollo-equipped workstations, the Chandler Limited GAV19T Amplifier Plug-In is an exacting emulation of the dual EL84-powered boutique guitar amp.
Inspired by British classics such as Vox and Marshall — and lesser-known gems from Watkins and Selmer — the Chandler Limited GAV19T Amplifier plug-in offers features not found on the original hardware, such as an FX Rack with an onboard noise gate, host-syncable lo-fi delay, Power Soak, and pre/post switchable EQ filter controls.
—Perfect emulation of the original dual EL84-powered Chandler Limited GAV19T circuit
—Fine-tune sounds with an onboard FX Rack that includes a noise gate, EQ filter controls, and host-syncable lo-fi delay
—Audition 75 different Recording Chains to match the perfect tone to the part
The Brainworx bx_saturator V2 Plug-In is amulti-band M/S (mid/side) processor that allows users to add saturation, drive, and distortion as desired. It offers Brainworx´s “True Split” crossover technology, which ensures that the mid and side channels are always perfectly in phase, yielding focused, large-sounding mixes and masters.
—Adds saturation and distortion to targeted frequency bands
—Increases the perceived volume of mixes without clipping
—Bring out individual frequencies of a mix without affecting others
—Mono Maker control helps define and punch-up low-end
The Brainworx bx_refinement Plug-In allows mixing and mastering engineers to remove the harsh, hard edges of their tracks without dramatically altering the character and tone of their source material.
Designed by mastering engineer Gebre Waddell of Stonebridge Mastering, the bx_refinement plug-in is based on a combination of time-tested approaches, modern techniques, and the latest science on the sensitivity of the human ear.
—Brings the sound of a tube-like analog mastering chain to DAW
—Softens high-frequency harshness in mixes and masters
—Injects saturation and presence to subtly enhance tracks
—Solo feature to hear exactly what’s being removed from the recording
Equator Audio D8 Monitors Central To Undercurrent Labs’ Production
Undercurrent Labs utilized Equator Audio monitors for the majority of their production work.
John Penn, CEO of Undercurrent Labs, producer, 3D sound designer and songwriter, insists upon the sonic accuracy of his D8 studio reference monitors from Equator Audio when working on projects.
Undercurrent Labs is the technology development arm of InspireMedia Interactive, an IPTV network with 3D audio and surround capabilities that streams both live video and VOD (Video-on-Demand) to any Internet enabled device.
As the company CEO, Penn wears many hats that include business development, managing client relationships, as well as directing and producing digital media projects that encompass sound design, audio for video, and music composition.
Further, he also consults with other firms that require subject matter expertise in mission critical audio and video applications.
“I use Equator Audio D8’s for every aspect of the production process,” states Penn, “from music composition for film and TV, 5.1 and 3D sound design, as well as mixing, mastering, and layback to video.
“Presently, I own seven D8 monitors. I usually configure five of the D8s in a planar L-C-R / L surround-R surround configuration for film and interactive projects, and I add two more D8s above the front L-R for the height channels when sound designing in 3D or mixing ambisonic projects.”
“I’ve been using the Equator monitors for the last year and, during that time, I handled a variety of projects, including ongoing development with 52WeeksMusic.com, plus a new film for which I’m sound designing and mixing.
“I’m also having an awesome time scoring the theme song for a new animation series created by my daughter, Maya Penn, and presented during her recent TED talk.”
Because the D8s emit a flat, non-hyped, full-range sound it lets the listener know that he/she is not missing anything. The D8s also have an overall clarity that speaks to the accuracy of the solid monitor build, which presents a punchy, tight bass response. Lastly, the mids have a distinct presence that complements the highs, which are never too harsh, even at higher SPLs.
“I believe many engineers will appreciate is how much better their mixes can sound using the D8s—because the D8’s don’t color or unnaturally boost the low end,” he adds. “Equally important, the D8s have a balanced response where you can still ‘feel’ the monitors.”
“Equator Audio provides a personal touch, which I really appreciate,” Penn reports. “Founder Ted Keffalo is very knowledgeable and provided a wealth of information about the D-series use in multiple monitor applications.
“He also proved to me how Equator was founded on technical excellence at an affordable price. And Equator Audio General Manager Marty Bradley provided a unique customer experience: from pre-sales to a flexible product delivery schedule as part of a larger order through RSPE Audio Solutions.
“Everyone at the company been easy to work with and very attentive to my particular audio requirements.”
Penn notes that he is pleased with how the D8s have performed on a diverse range of projects, from recording and mixing the hard-edged massive synth sounds of a Q-Bik Muz remix of Kinetic Motion to creating atmospherics for continuity in a film sound mix like Switching Lanes.
“The concentric sound puts me at ease in the sound field, enabling me to hear a pleasing, high resolution sound necessary to catch what other monitors miss—without fatigue and worries about translation of playback on mobile devices, theaters, cars, or at home,” he concludes. “My engineers and clients have also remarked how amazing these monitors sound.
“Prior to working with the Equator D8s, achieving this level of quality, full-range surround was nearly impossible at this price point. These monitors sound great, are very accurate, and represent exceptional value. I couldn’t be happier.”
Some of the most popular instruments in many genres of music are keyboards, so let’s look at some techniques to capture a grand piano, upright piano, Leslie organ speaker, digital keyboard or synthesizer.
This magnificent instrument is a challenge to record well. First have the piano tuned, and oil the pedals to reduce squeaks. You can prevent thumps by stuffing some foam or cloth under the pedal mechanism.
One popular method uses two spaced mics inside the piano. Use omni or cardioid condenser mics, ideally in shock mounts. Put the lid on the long stick. If you can, remove the lid to reduce boominess. Center one mic over the treble strings and one over the bass strings.
Typically, both microphones are 8 to 12 inches over the strings and 8 inches horizontally from the hammers (Figure 1). Aim the mics straight down. Pan the mics partly left and right for stereo.
The spaced mics might have phase cancellations when mixed to mono, so you might want to try coincident miking (Figure 1-A). Boom-mount a stereo mic, or an XY pair of cardioids crossed at 120 degrees. Miking close to the hammers sounds percussive; toward the tail has more tone.
One alternative is to put the treble mic near the hammers, and put the bass mic about 2 feet toward the tail (Figure 1-B).
Another method uses two ear-spaced omni condensers or an ORTF pair about 12 to 18 inches above the strings. With the ORTF stereo mic method, two cardioid mics are angled 110 degrees apart and spaced 7 inches horizontally.
Boundary mics work well, too. If you want to pick up the piano in mono, tape a boundary mic to the underside of the raised lid, in the center of the strings, near the hammers.
Use two for stereo over the bass and treble strings. Put the bass mic near the tail of the piano to equalize the mic distances to the hammers (Figure 1-C). If leakage is a problem, close the lid and cut EQ a little around 250 Hz to reduce boominess.
If your studio lacks a piano, consider using a software emulation of a piano. Some programs provide high-quality samples of piano notes that can be played with a sequencer or a MIDI controller.
Moving on, here are some ways to mike an upright piano (Figure 2):
A) Remove the panel in front of the piano to expose the strings over the keyboard. Place one mic near the bass strings and one near the treble strings about 8 inches away.
Record in stereo and pan the signals left and right for the desired piano width. If you can spare only one mic for the piano, just cover the treble strings.
B)Remove the top lid and upper panel. Put a stereo pair of mics about 1 foot in front and 1 foot over the top. If the piano is against a wall, angle the piano about 17 degrees from the wall to reduce tubby resonances.
C) Aim the soundboard into the room. Mike the bass and treble sides of the soundboard a few inches away. In this spot, the mics pick up less pedal thumps and other noises. Try cardioid dynamic mics with a presence peak.
Leslie Organ Speaker
This glorious device has a rotating dual-horn on top for highs and a woofer on the bottom for lows. Only one horn of the two makes sound; the other is for weight balance.
The swirling, grungy sound comes from the phasiness and Doppler effect of the rotating horn, and from the distorted tube electronics that drive the speaker. Here are a few ways to record it (Figure 3):
In mono: Mike the top and bottom separately, 3 inches to 1 foot away. Aim the mics into the louvers. In the top mic’s signal, roll off the lows below 150 Hz.
In stereo: Record the top horn with a stereo mic or a pair of mics out front. Put a mic with a good low end on the bottom speaker, and pan it to center.
When you record the Leslie, watch out for wind noise from the rotating horn and buzz from the motor. Mike farther away if you monitor these noises.
Rather than recording an actual Hammond B3 organ and Leslie speaker, you might prefer to use a software emulation of those instruments: an organ soft synth or sample and a Leslie rotating speaker plug-in.
Trigger the synth or sample by a MIDI sequencer or MIDI controller. You can automate the horn rotation speed in the Leslie speaker plug-in.
Synthesizer, Digital Keyboard, MIDI Sound Module, Drum Machine & Electric Piano
So far we covered acoustic keyboards which you pick up with a microphone. Now let’s look at electronic keyboards which you record with a cable.
One system uses a MIDI controller keyboard to trigger a software synthesizer (soft synth) in a computer recording program (Figure 4).
This is a whole subject in itself. Basically, the MIDI controller produces MIDI messages when you play it. These messages tell which keys you pressed, how hard you pressed them and when. It’s not an audio signal.
Figure 4. Recording and playing a soft synth using a MIDI controller keyboard.
A MIDI controller can be any electronic keyboard or drum machine that has a MIDI OUT connector. The MIDI signal triggers a soft synth in your computer. That synth can play any sound, such as a piano, bass, string section, and so on.
You record using a computer with an audio/MIDI sequencer program. After recording the MIDI events (the sequence), you can edit the MIDI notes to correct errors: change their pitch, timing, or duration.
Another way to record an electronic keyboard is to record its audio signal. For the most clarity, record the signal directly from the instrument.
In Figure 5, the top two parts show two ways to do it. Either plug the instrument into direct boxes, or into two phone-jack instrument inputs on your mixer or audio interface.
Figure 5. Three way to record the audio signal of an electronic keyboard.
Set the volume on the instrument about three-quarters up to get a strong signal. Try to get the sound you want from patch settings rather than EQ.
If you connect to a phone jack and hear hum, you probably have a ground loop. Here are some fixes:
Power your mixer and the instrument from the same outlet strip. If necessary, use a thick extension cord between the outlet strip and the instrument.
Use a direct box instead of a guitar cord, and set the ground-lift switch to the position where you monitor the least hum.
To reduce hum from a low-cost synth, use battery power instead of an AC adapter.
A synth can sound dry and sterile. To get a livelier, funkier sound, you might run the synth signal into a guitar amp, instrument amp, or power amp and speaker. Mike the speaker cone a few inches to a few feet away (Figure 5, bottom part).
You might mix the microphone’s signal with the direct signal. If your recording system has a polarity button in the mic’s channel, flip the button in and out, and choose the position where you monitor the best sound.
Does the keyboard player have several keyboards plugged into a keyboard mixer? You may want to record a premixed signal from that mixer’s output. Record both outputs of stereo keyboards.
A member of AES and SynAudCon, Bruce Bartlett is a live sound and recording engineer, microphone designer (http://www.bartlettaudio.com), and audio journalist. His latest book is Practical Recording Techniques, 6th Edition.
Lightning Boy Audio Introduces 1401 All-Tube Stereo Microphone Preamp
New preamp is a combo or the 1401 stereo microphone amp and the Ghost Box
Lightning Boy Audio (LBA) has announced the availability of the 1401 stereo microphone amp and the Ghost Box, a combination that forms a versatile all-tube stereo microphone preamp.
Through the use of high quality components such as NOS paper in oil capacitors, NOS tubes, Carnhill iron and the careful layout of point-to-point wiring, the 1401 Preamp (short name) and Ghost Box command a stereo image with high fidelity.
The preamp was designed to serve as the output amplifier for the LBA 1401 Analog Plate Reverb, but with the addition of the Ghost Box, it’s capable of serving most tracking needs in the studio. The 1401 Preamp’s front panel offers gain and passive treble controls for two channels, while the Ghost Box provides phantom power, phase invert switches and passive low-cut switches for two channels.
The Ghost Box offers a new take on phantom power, which utilizes a 6X4WA Rectifier tube to create 48-volt DC and an NE-16 neon regulator tube to regulate voltage. It brings out a unique tonal variation in any condenser mic powered by it.
As with all LBA gear, no solid state components are used. The 1401 Preamp can be ordered directly from the company website, lightningboyaudio.com.
Reverb. It’s use goes through cycles from a lot to almost none, but you’ll usually find at least some reverb-type ambience used in every mix. The problem is that you can’t really tell much of a distinction between the different types of some inexpensive plug-ins or boxes.
There are five primary categories of reverb, all with a different sonic character; three of these are actual acoustic spaces, one is an analog way to reproduce one, and one is not found in nature but can really sound cool.
The reason why there’s a difference is that just like everything else in music and audio, there are many paths to the same end result. You’ll find that every digital reverb plugin or hardware unit provides its own version of these sounds.
Hall: A hall is a large space that has a long decay time and lots of reflections. Sometimes there’s a subcategory of the hall reverb called “church,” which is just a more reflective hall with a longer decay.
Room: A room is a much smaller space that can be dead or reflective, depending upon the material that the walls, floor and ceiling are made of. It usually has a short decay time of about 1.5 seconds or less.
Chamber: An acoustic chamber is a dedicated tiled room that many large studios used to build in order to create reverb (Figure 1). Phil Spector’s “Wall of Sound” was built around an excellent acoustic chamber at Gold Star Studios in Hollywood (long since closed unfortunately), for example.
The acoustic chambers at Capitol Studios, designed by Les Paul himself, still have a reverb sound that is revered by mixers everywhere. Other common artificial spaces used as acoustic chambers include showers and stairwells.
Plate: A plate is a 4-foot by 6-foot hanging piece of sheet metal with transducers attached to it that many studios used for artificial reverb when they couldn’t afford to build a chamber (Figure 2). The first plate reverb was the EMT 140, developed in the late 1950s, and is still held in high esteem by many mixers for its smooth sound.
Non-Linear: The non-linear category is strictly a product of modern digital reverbs as the sound isn’t found anywhere in nature. While natural reverbs decay in a rather smooth manner, once a reverb is created digitally, it’s possible to make that decay happen in unusual ways.
The reverb tail can be reversed so it builds instead of decays, or it can be made to decay abruptly, both of which makes the decay “non-linear.” This preset was a popular mixing effect used on drums during the 80’s when the feature first became available on the AMS RMX 16 digital reverb (Figure 3).
As to which reverb category to use, that’s strictly up to the taste of the mixer and how he sees it fitting with the song. Many mixers might always use a room or a chamber on drums, a plate on vocals or guitars, and a hall on strings or keyboards, while others may do just the opposite.
Many might refine the sounds of each and finally settle on a few that they feel always work in a particular situation with a certain instrument or vocal. It’s all up to experience.
All of these reverbs can be modeled using what’s known as a “convolution reverb” that uses an quick burst of audio energy (called an impulse) to excite the room or device, which then allows it to sample its parameters. Examples of convolution reverbs include the Audio Ease Altiverb and Avid’s TL Space.
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. Get The Mixing Engineer’s Handbook 3rd edition here.
This page has been viewed 0 times
Page rendered in 0.8530 seconds
Total Entries: 20072
Total Comments: 1997
Total Trackbacks: 0
Most Recent Entry: 05/27/2016 12:21 pm
Most Recent Comment on: 01/19/2012 02:32 am
Total Members: 4923
Total Logged in members: 0
Total guests: 2
Total anonymous users: 0
Most Recent Visitor on: 02/10/2012 11:04 am
The most visitors ever was 774 on 02/08/2012 02:19 pm