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Getting your head around a new console, especially one that has custom features can be a challenge. I recently sat in on a session in a room that had a Neve 8078 console that had been completely refurbished and hot rodded. This one can operate as an inline or split desk and the rear bus is purposed as a second stereo bus. It also had other custom features that are purposed on existing switches. For instance, the Direct button sends the 100mm fader to the multitrack buses much like the Float button on an SSL 4000 series console. Lots to remember!
When trying to get up to speed, the perfect scenario is to watch someone who is an expert operate the desk in session who is welcome to questions, which was the case with me. But what if you are thrown into the deep end by yourself? You would expect the studio to have an able assistant who can help but you still need to have your questions ready so you can get up and running quickly.
For starters, I find myself gravitating towards the basics first then more advanced controls. For instance, knowing how to switch between recording and mixing workflows is something you’ll need to know, but where are the buttons? Are the controls global, per channel or both? Here’s a breakdown of the top 10 essential console features you’ll need to know when working on a new desk.
Control Room Volume
Source Fader Control and Flip (if available)
Global Controls (flip, mute and solo clear, grouped solo safe etc)
Multitrack Bus Pan
Aux Send Masters
This is just a start but should get you through a session. Like anything else, say learning to drive a car, you feel clunky at first but soon get used to the controls. Knowing where the controls for the ten functions above reside should get you on your way to Ninja status on a new desk.
I recently had a call from Simon Coté at AudioPlus Services who asked if I knew of a system that could bass manage an 11.1 speaker array. Good question, as there isn’t a lot of turnkey surround gear out there beyond 7.1.
Although it wasn’t necessary for the search, I was curious about the setup of the speakers. The system would be fed 11 channels of discrete audio and consist of Focal Twin Solo speakers setup in a standard 7.1 array; LCR (22°-30°), LS/RS (90°-110°) and LB/RB (135°-150°). The other four channels would be Focal Twin Solo 6 monitors setup as two high front speakers and two high back speakers.
I started my search for a bass manager by calling my friend and engineer David Rideau who as my luck would have it, was at GenAudio’s excellent 5.1 room in Los Angeles for a session. Genaudio is all about surround, both discrete and simulated and if anyone would know about multichannel bass management, they would. Dave fired the question to GenAudio‘s Senior Mix Engineer Greg Morgenstein who came up with Meyer Sound’s Galileo Speaker Management System as an option. I let the guys get back to work and searched on Meyer Sound’s Website for the specs.
Unfortunately, Galileo has only 8 inputs per unit which is perfect for 7.1 but not 11.1. Although you can gang Galileo systems together, you can’t manage inputs from both units to a single output. My search continued on Meyer Sound’s Website where I found D-Mitri, a versatile, high-resolution audio system that does speaker management plus much more. D-Mitri is a scalable system where you can design your own analog and/or digital IO, include a recorder, do advanced speaker management, matrix mixing and routing and choose enough DSP for your setup which can include up to 144 outputs.
Communication between D-Mitri’s many boxes is via Ethernet, so a very large system can be setup with a small cabling footprint. I confirmed that D-Mitri was the system for the job through a quick call to Brian Bolly from Meyer Sound’s Cinema division. Brian walked me through D-Mitri’s options which were all very cool and although the system had more horsepower than I needed, was perfect for the job.
I called Simon back with my findings and found that in his own search, had found another option; the Trinnov Magnitude32 Room/Speaker Optimizer. The Magnitude32 is another scalable system but takes the single box approach where you can add expansion boards for more IO and get up to 16 inputs and 32 outputs. Magnitude32 has enough DSP power for Mixing Matrixes, Manual FIR filters, Parametric EQs, Graphic EQs, gains, trims, RMS meters, manual delays, Bass Managment, 4-way active crossovers and more.
The process of finding the right manager for the job was educational for me. I started the day not knowing much about advanced systems like these and by the end of the day had two new systems in my lexicon for advanced bass management and much more. I love audio!
Unit Audio UNIT Passive Summing Mixer
by Brandon Hickey
Unit Audio makes a small array of hand-build utility products for recording and mixing. Their primary product line offers passive analog summing mixers for DAW-based workflows. The benefits of analog summing is that combining electrical signals through an analog mixing buss sounds different from summing inside of a computer. Though DAW-based mixing offers the advantages of easy recall and automation as well as plugins which are more cost friendly than outboard hardware, there are also distinct advantages to analog mixing. Consoles usually offer greater headroom than DAW mixers, and in many cases color the sound with a unique flavor.
Unlike summing systems like the Dangerous Music 2-Bus which offers a fully active circuit path, the UNIT falls in line with devices like the ROLL Folcrom which features no active circuitry. Passive mixers combine signals by using a network of resistors to sum them together. The resulting output is very low in level, so a microphone preamplifier is necessary to restore it to line level before hitting the mix-down deck. One supposed benefit of this type of strategy is that the mic preamp adds coloration to the signal. Naturally this is a point of debate. For example, how many engineers plug a mic into a mic pre, then to a re-amp, then a DI, then to another mic pre to add flavor to the signal? Once a signal hits line level, why not keep it there, right? That is how I felt too, before I tried the UNIT. After using it, I haven’t necessarily been born again, but I will say that the results were not quite what I had expected.
The UNIT is 16 x 2 having two DB-25 connectors, each accepting eight balanced signals, and outputting to a pair of balanced Neutrik TRS connectors. UNIT Audio also produces an 8 x 2 mixer with all TRS connections, and may have XLR output connectors upon request. Several of the models have a pair of switches that can break a stereo pair of input into two mono sources, so that kick and vocal, for example, can be inputted individually without having to waste a stereo pair with dual mono signals.
The Sum Of All “Hears”
The primary material for this listening test was a jazz trio which had been tracked to tape, and transferred to Pro Tools. I mixed in the box using only plugin-based effects, then produced stems of drums, kick, snare, bass, sax, and a few effects stems. I then combined those stems together using the Pro Tools mixer, the software mixer in Harrison’s Mixbus DAW, the UNIT, and two analog consoles (SSL 4000E and an API Legacy Plus), with faders set at 0 dB VU.
It is important to note, at this point that the sound of the Xicon resistors and Neutrik connectors will play a role in the sound, but maybe not as significant as the roles played by the DA converters outputting the DAW signals to the box, the sound of the mic preamp gaining the signal back up, nor the AD converters feeding the mixdown deck. In fact, even the cabling at either end will color the sound, and perhaps even more so than the resistor network. All that said, subjectively grading the sound of the UNIT itself is nearly impossible. Despite that, my goal was to color the mix minimally and merely take advantage of the added clarity, enhanced stereo image, and tighter bottom end that the UNIT promised. I used Monster snakes at the front end, Canare Star-Quad cable with Neutrik connectors on the back end, and fed the clean, quiet, Sound Devices USBPre2 for mic pre and A to D conversion. For the sake of convenience and consistency I used the D to A converters of a Digidesign 96 I/O.
Referencing all mixes through a number of monitors and headphones revealed subtle differences. Color and frequency response changes were minor, but there were also a few big things that really stood out. The most profound factor was the way the mix sounded when things got really fast and busy. For example, there were moments in the tune when the sax started playing higher notes at a point in the groove where the bass was making fast transitions and a drum fill took place. With all of that happening at the same time, the Pro Tools mix quickly turned muddy. Harrison’s Mixbus stayed clearer through these dense sections than Pro Tools, but still had a tendency to lose definition when compared to any of the analog mixes. The SSL sounded more like the Harrison software mixer though subtly clearer on quick fills. Both the API and the UNIT shared a similar character with each of them producing a sound which was like a pleasant, natural, harmonic distortion which accented the attack of each drum hit, propelling it through the mix. At the same time, this sound was persistent throughout the mix, creating an unnatural high-frequency boost.
On one hand, I would say that having heard the comparison, the way complex signals combined through the UNIT was preferable to Pro Tools. The API shined, adding all kinds of warmth and flavor through the midrange and low frequency range. That said, the sound of either of these solutions is far from transparent, and you would have to be careful to choose the right pair of mic pre’s to compliment each mix. Even then, you’ll never get a colorless sound, and may find yourself EQ’ing around the mic pre. Also worth noting was the change in stereo image across these different mixes. Relative to the Pro Tools sum, Mixbus and API both widened the mix a bit, with the SSL mix being the widest of all. The UNIT mix sounded close to Pro Tools, but if anything, it seemed like the midrange was a bit narrower.
When it comes down to it, summing outside of the box is undeniably popular, and engineers are looking for that sound. Some passive summers, like the Shadow Hills Equinox include the makeup gain pre-amplifier which drives up the cost significantly. Meanwhile, the UNIT is about a third of the price of the ROLL Folcrom, and does the same thing, minus some unnecessary routing switches. While this product is not going to be as clean as an active summing amp, it does provide an interesting sound, which, for many musical applications may be just the kind of thing you are looking for. Certainly it is a different character than a mathematic mixer, and if your mixes are stuck in a rut, this might be just the thing to help you think outside of the box.
COMPANY: UNIT Audio
PRICE: $149 – $399 plus shipping
When mixing certain kinds of percussion, kick drum or low tom, you can use this old trick to add some extra bottom end. I’m triggering my BOOM channel from dedicated LFE channels but you can do this from any audio source, mono, stereo or whatever.
1. For each channel you wish to add BOOM, make a mono aux channel, mute it and instantiate a signal generator on the first insert, and a gate on the second insert (I use a Waves C1 gate)
2. Set the Hz value of the generator to 40Hz (you can fine tune this later)
3. Set the gate open value to about -20 (see pic of Waves C1 gate below for reference)
4. On the audio track you’re triggering the BOOM from, create an aux send to any bus, raise the send fader to unity gain and make it pre-fader. This way, later on when you change your mix, the BOOM channel will still trigger consistently.
5. Set the key input of the gate on the BOOM channel to the same bus sent from the trigger channel
Now you’re ready to go! Play your audio track and fine tune the gate open value until the BOOM is happening just on the accents. If the gate causes clicking you can play with the attack and release to make it more musical.
I’m triggering from an LFE track which is all low end, but you can make any channel work by duplicating it, sending the trigger from the duplicate, and putting an extreme Low Pass filter on the track. This way you’re only getting the bottom end going out of the trigger channel. Ping me on Facebook and let me know if it’s working for you. Overusing this trick can muddy the bottom end of your mix but if you pick your tracks carefully, it can be a great way to bring out the bottom octave of a mix.
In part 1 of my vocal smoothing feature, I outlined the initial steps you can take to make a lead vocal track sound smooth, understandable and natural. Once I apply the plugins and get my hardware sounding great, I move on to automation to put the finishing touches on the track. I’m using Pro Tools but the techniques here can be used in any DAW.
For starters, I’ll write an automation pass with the vocal track’s fader at a fixed level that works as a starting point. Do this by opening the Automation Window (Command + 4 on the numeric keypad), Select VOL under Write Enable and WRITE ON STOP with the arrow pointing in both directions. Set the vocal track’s automation selector to WRITE then use the space bar to start and immediately stop the playhead. You’ve just written the volume of that fader at that level for the entire song. With this done, if you make automation moves in TOUCH, the fader will always pop back to this level when you let go – very handy.
Once I flatline the fader, I’ll manually write in dips and tucks on problem areas in the track’s Volume View. For example, I’ll listen to the track and find any problem areas (remaining sibilance, plosives) and dip those at least -3dB (sometimes more) with V cut in the Volume line. Do this with the grabber tool (Command + 4). Hover over the Volume line with the Grabber and click to create break points. I’ll break the line bit before and a bit after the area, then put another break dead center between them and drag it down paying attention to the -dB amount as I go. Depending on how much you have zoomed in on the track, you’ll get a hang of the overall size of the dip after you do a few and listen back. I’ll use this technique throughout the entire vocal writing in dips wherever sibilance, unpleasant volume shifts, or large breaths need to be reduced. This can take a while but it’s worth it.
For me, this manual approach is preferred to addressing the dips with a controller because of the delay caused by the plugins. Plugin latency is handled automatically by Delay Compensation so it’s correct to your ear, but if you make fader moves in real time, they’ll be late. For this reason I stick to manual writing with the mouse for the first pass and keep the fader moves for more global trimming.
Once all my cuts are in, I’ll make a VCA Master fader to do my final moves. You could make moves on the vocal track itself in Touch/Trim, but I’d rather leave my dips untouched for fine tuning and have the more global fader moves on another track.
To get the VCA to control the vocal fader, you’ll have to group it. Select the Vocal track, push Command + G, then name the group anything you’d like. Then go to the VCA Master fader and select that group. Now the VCA master is controlling your vocal fader and you can write automation. Write the VCA flat at unity gain (0) for the entire song as above, then put the VCA Master in TOUCH and start making your moves. Here’s where you have to be aware of the delay caused by the plugins. I’ll listen to the track and find spots that need smoothing, then I’ll make the move early to compensate. It will take a few passes to get this right but once you have a feel for it, you’ll be making moves with minimal re-dos. Once you go through the entire song, you should really start hearing the vocal sit down in the mix, sounding very smooth and natural with each word being heard. Happy Smoothing and ping me on Facebook with your own mixing tips.
One thing that brings a high level of professionalism to any mix is making your vocal sound smooth, understandable and natural. A compressor goes a long way to make this happen but it also brings the blemishes to the surface. As you limit dynamic range, you start to hear bad edits, hard syllables, sibilance and the difference in takes if you made a comp. Each one of these pimples takes a special skillset to make right.
Before I get into individual skills, let’s talk about the vocal chain. I’m going speak from my own perspective as a mixer so my gear may differ from yours, but the concepts remain the same. I have a hybrid setup with various analog tube gear and plugins from UAD, SoundToys, VSL, Slate, EastWest, Avid, Steinberg and more. My vocal chain plugins vary greatly depending on the singer, but I always end the chain by leaving my converters and going through a Millennia STT-1 and then into a Dangerous 2-Bus and BAX EQ before it splits back to a Dangerous Monitor ST and back to my Lynx converters for burning a 2 mix.
The STT-1 has tube or solid-state sections for the EQ and compressor and input section. I will choose more tube over SS if the vocal needs some taming down from any digital harshness and I generally use the four EQ bands as a way to balance the tone in the 100Hz range, at the fundamental (depends on the vocalist), 4k to 5kHz and 8Khz and up. The compressor is working between 5 to 8 dB of gain reduction depending on how dynamic the part is.
I’m also giving the vocal a lot of plugin and automation love before it gets out of the box. I’ll always tune before I process so that plugin is first in line. Lately I’ve been using Slate VCC plugins on all my channels and VCC bus plugins on various groups, or I’ll sometimes hit the track with a UAD ATR-102 plugin for some tape simulation – or both. Next comes a utility compressor such as the UAD Fairchild or if the track is especially dynamic, I’ll go deep with a FET/VCA style compressor like an 1176, DBX 160 or Fatso Jr. plugin. These can sound “pumpy” (with hard-knee and apparent gain changes) so you have to be wary of hitting the track too hard here. But with just the right setting, you can tame the peaks with a good amount of transparency. I’m often barely moving the needle here, just getting the peaks.
Next, I use at least one, and often two or three de-essers depending on what the track sounds like. I prefer the UAD Precision De-esser and UAD bx_digital V2 which lets you dial in the frequency to cut over the full range of audible frequencies. With the V2, you can use the Listen mode to find an offending tone by grabbing a frequency knob which solos just the frequency you set it to, boosting it at a very narrow bandwidth. You can then tune in on the harsh tone to get the frequency you need to plug in to the de-esser. I’ll then shut off that band so I’m not using the EQ, and plug the number into the de-esser (see V2 pic above with de-esser outlined in white.) Then you just dial out the amount to cut which is measured in -dB steps and is reflected by the GR meter. There is also a handy solo button so you can hear what’s going on when the GR meter kicks in. It’s very easy to use and sounds great.
Anyone else seen this? It’s a list of track presets in Pro Tools 10.1.2. After viewing this, I was unable to call it up in any other newer version. Very strange but also a great feature! If you’ve seen this, ping me on facebook and let’s chat about it.
Endless Analog has released a new range of products aimed at those wanting to get into CLASP without breaking the bank. The new products are CLASP 8, CLASP 16, Machine Matrix, and Machine Matrix I/O. The 8 and 16 are scaled down versions of the original CLASP so those with more limited analog track counts, don’t have to pay for 24 channels. The Matrix provides 8 channels of audio. The Matrix I-O provides an additional 8 channels per unit. All the gear works together to allow users to jump between tape machines, and use 8 (or more) channels at a time, depending on how many I/O boxes are purchased.
CinemaCon opened this week in Las Vegas and brought with it the release of two new Surround Sound creation packages for film mixing. CINEAUDIO with AstoundSurround is a software solution allowing theaters to play back 11.1 or more on existing 5.1 systems with no additional hardware. Dolby’s new ATMOS on the other hand is a scalable system that is object rather than channel based. It involves sophisticated panning executed during the mix that is encoded, then decoded on site. The video below explains it all.
Also in the game, but not exhibiting at CinemaCon is SRS Labs’ who just released MDA (Multi Dimensional Audio). MDA promises to represent sound sources as objects in space just as they are in the real world, without regard to the number of channels or speaker locations.
“In the theater, the CINEAUDIO cinema processor unit integrated with AstoundSurround® for Cinema software takes the multi-channel audio printmaster with discrete channel inputs (e.g. 7.1, 11.1 or more) in real-time and renders the same multi- channel audio experience in any existing 5.1 multi-channel audio equipped room, maintaining all panning information (including elevated audio channel information).” READ MORE
For the first time, Dolby Atmos introduces a hybrid approach to mixing and directs sound as dynamic objects that envelop the listener, in combination with channels for playback. Dolby Atmos enables adaptive rendering to ensure that the playback experience is as close as possible to the creator’s original vision in any given environment, irrespective of the specific speaker configuration in the playback environment.