“Thank you, this is a huge win for me. I appreciate you and your company.”
“Thank you, this is a huge win for me. I appreciate you and your company.”
“You got to be kidding me!!!! Man I have never won anything. I’m stoked!”
We’re stoked too, Paul! Enjoy your new speakers.
At NAMM this week, Avid released its newest iteration of Pro Tools software. Let’s dive in a take a look at some of the new features.
This update will be especially useful for songwriters and producers. It includes retrospective MIDI recording, easier transposing/trimming/velocity changes, and useful shortcuts. Learn more from Avid here:
My personal favorite addition to Pro Tools with this update is their Track Presets feature. This allows you to save tracks (their settings, plugin chains, even sends) as presets to be recalled in a variety of different ways. You can even create folders to store these presets in for better organization. This could be something like “Vocals Presets, Guitars Presets, Delays Presets, etc.” for example.
Imagine creating a reverb send just by clicking “new send” from the source track, and then choosing one of your reverb track presets to instantly create your favorite reverb chain, including the reverb plugin, EQs, and compressors. Well now you can! How cool is that?! Watch Avid’s Pro Tools Track Presets in action on Avid’s YouTube channel:
Avid made some major changes to mix view. The first I’ll cover has to do with playlists and comping. New shortcuts for comping playlists allow us to select one “primary” playlist to send our comps to, which means we can now comp takes in waveform view, quite a space saver when you’re comping takes with multiple tracks such as drums.
Secondly, Avid has added an EQ window similar to Logic’s, with the exception that Avid’s EQ window works with third-party plugins! As someone who bounces back and forth between Avid and Logic, this one jumped out at me. I’ve always thought Logic’s EQ window was a neat feature but that it’s sort-of wasted by its limitation to Logic’s native EQ. Avid did this one right! On top of providing support for third-party plugins, the EQ window in Pro Tools shows all the EQ plugins on the track, combined. So you’re seeing a culmination of your entire signal chain in visual format. Pretty useful stuff. Learn more from Avid:
As has been rumored for several months, Pro Tools 2018.1 included support for iLok’s new iLok Cloud. This means Pro Tools 12 and Pro Tools HD no longer require a physical iLok dongle to be present in order to work. As someone who has lamented the loss of a USB port since, well, always, I’m not over-the-moon about iLok Cloud… yet. This was a huge step and we should be thanking Avid for including this capability with Pro Tools moving forward, but unfortunately, we’ll probably be waiting for many of our third-party plugins manufacturers to come around for a while longer. So alas! Depending on the third-party plugins you use, you might be stuck with the iLok dongle a little while longer. That being said, this is certainly still cause for celebration. And on that note and all of the above, thanks to Avid for listening to your user base and working hard to implement all of these incredible new features!
Now! As of today you can download the newest version of Pro Tools 12 or Pro Tools HD as part of your Annual Upgrade Subscription or purchase and download it as a new copy:
Need to renew your Pro Tools 12 or Pro Tools HD subscription? Grab them by clicking one of the links below:
If you’re heading to NAMM this week, make sure to stop by and see us at the LaCie booth (Level 2,
Booth #17905). We’ll be hanging out there throughout the week, answering questions and such. We’re offering a special discount on LaCie products if you swing by and ask!
Guitar Amps: Tu-be Or Not Tu-be
“Whether ’tis nobler in the mind to gig with a tube amp or a solid-state guitar amp and suffer the slings and arrows of capacitor-sniffing tone snobs”
It’s pretty much a given that your more militant guitar tonemeisters prefer tube amps to solid state. According to Professor Alexander Dumble of Dumble* amps fame, “The fragile harmonics live better in tubes than the crystal lattice of transistors.”
To us mere mortals, the simple truth is that the main difference between tubes and transistors lies in their behavior as amplitude increases. Tubes distort gradually with volume exhibiting a natural compression, whereas transistors perform consistently at increasing volume until they go into distortion. When a transistor clips, it chops off the top of the wave (and throws it to the floor rather harshly), which is very unpleasant sounding. Of the two, tube distortion is much more natural and pleasing to our ears. Tubes distort in much the same way our voices do when we go from talking loudly, to a yell, to a scream. Solid-state amps can scream without distorting, but when they do distort, they lose their voice, in a manner of speaking.
Perhaps the biggest misconception in analog circuit design is that tubes sound inherently warmer than solid state. In truth, transformers are more responsible for the warmth of a sound than tubes. In fact, solid-state devices can sound quite warm, depending on the circuit topology surrounding them (think Neve 1073). However, when it comes to distortion, cranking a tube amp gives us the “right” sound for guitars. Tube designs sound better to us because there are fewer components in the signal path than solid state. Of course, the number of components in the signal path is more important when using a solid-state device for recording as opposed to instrument amplification. A transformerless solid-state preamp is as transparent as it gets. Transformerless tube designs are expensive and very hard to do.
The Right Amp for the Job
The type of amp you prefer has a lot to do with your style of music. Certain types of metal and thrash, particularly fast playing, seem to do better with tube amps, where high volume is the order of the day, and distortion comes from the tubes themselves. On the other side of the musical spectrum, jazz works well with solid-state amps, especially if you want your clean tones to stay clean regardless of how much harder you may attack the strings (tubes will distort with heavier attacks). The solid-state Roland JC-120 Jazz Chorus became the de facto jazz amp starting in the mid ’70s. Again, it’s a point of personal preference.
In a choice between a tube and solid-state guitar amp, apart from the style of music you play, the major consideration is whether you’re gigging or recording. The harmonic richness of tube amps is generally preferred in the studio, where the goal is the illusion of larger-than-life sound. The advantage of solid-state amps rests mainly in live performance. They’re lighter in weight (easier to carry to a gig) and not subject to vibration damage in transport the way that tube amps are. They also don’t require any warm-up time, so if you’re late to a gig, you can just plug in and play. This is of particular note if you’re gigging in the winter and your tube amp is in the back of a pickup, or left in a van overnight. Tubes respond poorly to having their heaters lit when they’re freezing cold.
FX Math: Analog Plus Digital Effects, Carry the One, Cancel the Tubes
Unless your effects setup is all analog, or a pricey rack unit (such as Eventide’s Eclipse) patched in through an effects loop, playing a gig with a tube amp can be an exercise in diminishing returns if you’re using one of the prosumer digital multi-effects pedals between your guitar and the amp’s input. Apart from the digital processor taking all the depth out of your tube amp (try an A/B comparison if you don’t believe it), you’d be better off with the high-gain-before-clipping response of solid state. Besides, why go through the care and feeding hassles of gigging with a tube amp only to make it sound solid state?
On the other hand, if you’re a purist playing blues, blues-rock, and certain styles of classic rock, the response of a tube amp and pure analog effects pedals will offer the preferred sound. It’s simply a matter of picking your battles. That is to say, if it’s a high-profile gig, then you’ll want your best gear. If it’s a roadside bar where people will be doing their damnedest to talk over you, you may want to go lightweight and solid state.
* Dumble amps were made by H. Alexander Dumble in the late ’70s to ’80s. The preferred amp of guitarists such as Eric Johnson, Robben Ford, Carlos Santana and Larry Carlton, each Dumble was hand-built and voiced to the playing style of its eventual owner, who waited three to five years for their amp. Only 300 made, Dumbles sell on the vintage market for anywhere from $25,000 – $50,000, and as high as $160,000.
Cables pretty much don’t do anything but pass signal. As a result, the materials and properties of those materials are turned into features, and subsequently the subject of marketing-speak and endless screeds of pros and cons. For example, oxygen-free copper (OFC) is supposed to have better conductivity, and therefore “sound” better than standard copper wire. The problem is, that’s kind of hard to prove unequivocally.
According to Roger Russell (former director of acoustic research at the McIntosh Laboratory) who is quoted often in online debates regarding the benefits of oxygen-free copper, better sound through increased conductivity is not a feature of OFC. According to Russell, “Highly refined copper (C10100 Oxygen Free Electronic or OFE) with silver impurities removed and oxygen reduced to 0.0005%, has only one-percent higher conductivity than C10200, known as Oxygen-Free (OF). Its conductivity rating is no better than the more common C11000 grade. However, C10200 has a 0.001% oxygen content, 99.95% purity and minimum 100% IACS* conductivity.” He further states that C10100’s improved conductivity is insignificant in audio applications.
You’ll never take me alive, coppers
Naturally, such a statement causes manufacturers of audiophile cables to foam at the contacts. One opposing argument is that Russell only mentions two types of copper, and there are more types of OFC. Beyond that, there isn’t a specific, universally agreed-upon set of manufacturing specs and processes to make OFC, and nothing quantified for audio applications.
So, when manufacturers tout OFC, are they taking you for a ride? Not really. Keep in mind that Russell was attempting to debunk the notion that OFC speaker wire did nothing to enhance sound quality in home stereo systems other than separate audiophiles from their trust funds. However, his main thrust was conductivity alone. Perhaps there’s more to the story than a simple conductivity rating? Some cable designers say you need to look at the atomic level.
One possible explanation for the perceived improvement of sound with oxygen-free copper cables is the fact that copper is crystalline in nature, with each crystal being a boundary that electrons have to cross. Typical high-purity electrical grade copper has approximately 1500 crystals per foot, which must be crossed by signal being transmitted through the cable. Now, think of the impurities in copper wire like knots in a rope, around which electrons have to navigate. Travelling around these impurities can cause phase shift and distortion that degrades sound quality. OFC is like pulling the knots tighter, which reduces the hurdles electrons must travel through, thus reducing distortion and phase shift. Another take on OFC is that the process of electrolytic refining also removes other impurities, most notably, iron, which can cause resistance. Be advised that the length of cable for these effects to be noticeable is up for grabs—some say over 50 feet—which for the home stereo is not an issue, but if you’re wiring a recording studio, oh yes it is.
Another perceived benefit of oxygen-free copper is due to the susceptibility of copper to corrosion when exposed to air. If you’ve ever had the pleasure of removing an old car battery, you know that copper exposed to oxygen corrodes battery cables, and stops conductivity dead in its tracks (electrons go in, but they don’t come out). According to tests, oxygen-free copper also runs cooler than other conductors. It’s more resistant to shorts, more durable and long-lasting, and is far less likely to corrode, due to the reduced oxygen content—at least, that’s the common wisdom.
Oxymoron no more
In light of all this, it becomes clear that the reason for OFC is due to consistent signal transfer in precision applications, longer life, greater reliability, and better performance over long cable runs. As such, if you’ve just installed a Rupert Neve Designs 5088 console in your studio to the tune of six figures, you’re going to want your cables to carry that tune all the way to number one on the Billboard charts. And, as Mogami states, there is no single magic bullet to creating superior cables. Rather, a combination of several factors, of which their years of research has determined that OFC copper is one. That, and the fact that cable is not meant to modify or alter signal in any way. Its job is to pass audio as transparently as the laws of physics will allow.
Finally, if you’re one of those engineers who likes to tweeze every microgram of sound out of your equipment, oxygen-free cable certainly isn’t going to hurt.
* IACS: International Annealed Copper Standard
Now that more mixers are moving inside the box, we’re going to focus on a mix-bus plug-in setup that will give you excellent results. Since plug-ins now model the behavior of analog equipment, a good starting point would be to set up a mix bus the way you would work in a studio with a large-format console going to tape. If you have console and tape emulation plug-ins, you can set up a mix bus to model a widely used studio setup that spawned a thousand hits.
Start with a Clean Slate
In our example, we’re going to use the Slate Virtual Mix Rack with a console mix bus module (the gray one that says “MIXBUSS”). You can set the console model to one of 6 emulations, including two famous SSL desks, an API console, the legendary Trident A-Range console, or a custom RCA tube console. Next, pull up a UAD model of an SSL 4K bus compressor. From there, insert a tape emulation plug-in, from either Slate or UAD, depending on which you have or prefer. The settings are those that would normally be used for a 2-track master. For example, the Slate tape emulation plug-in gives you the option of choosing between 16-track and 2-track machines, two models of tape; Ampex 456 (FG456) and Quantegy GP9 (FG9), and 15ips or 30ips tape speed. For a master mix, select 2-track, FG9 tape for more clarity and punch, and 30ips tape speed to preserve transients. Set your console emulation and tape emulation to operate at nominal levels (0dB UV average).
Master and Compressor
Now that we have our large-format console mixdown emulation, how do we set the SSL bus compressor? The trick is to barely touch the compressor. Set the ratio at 4:1, slowest attack (30ms), fastest release (.1s), and set the threshold so that the needle just wavers slightly. Gain makeup should be around 1dB. You can stop there, or you can make things more interesting by adding other compressor plug-ins for their tonal quality. Here is where you can begin to craft your own final mix sound. For example, after the tape emulation, you might choose to use the Slate FG Red, which emulates the Focusrite Red 3, or perhaps a Sonnox Inflator. In either case, you’ll hear another increase in body.
The one recurring theme is to barely hit the compressor. The advantage of this approach is that you can boost your level one or two dB at a time with each instantiation and not overload the mix bus.
If you hear a resonance you’d like to tame, or a magical frequency you want to bring out, you can insert an EQ after the compressors, such as the UAD Manley Massive Passive. Just make sure it’s a high-quality EQ plug-in.
Expanding Your Horizons
The last component in the signal chain is a limiter, which is where you get your biggest volume boost. Some excellent choices would be the FabFilter Pro L, UAD Precision Limiter (if you’re looking for very transparent gain), and the McDSP ML4000 multiband limiter. While McDSP now has an 8-band limiter, the beauty of the ML4000 is that it’s now much more affordable and enables you to shape the tone of the entire mix via its expander modules. The expanders push more signal into the peak limiter section, filling out your mix in ways that the compressors don’t. (It’s actually quite surprising.) The ML4000 will give your mix a competitive level in relation to commercial releases, enabling you or your clients to hear what the mix will sound like after mastering—or, you just may wish to leave it that way and send it off to mastering.
One of the techniques that make up the “Abbey Road Sound” is referred to, colloquially, as the “Abbey Road Reverb Trick.” Before we discuss the how-to, let’s talk about the results. If you listen to plate or chamber reverbs in their natural state, there’s a lot of information going on throughout the frequency spectrum. In short, the lows make mud in a mix, while the highs can result in oddly distracting, unnatural reverb tails (think ’80s-style ballad reverbs). In reality space, we don’t often hear such reverb decays, and certainly not as distinctly as when exclusively assigned to a particular sound in a loudspeaker mix.
You Do You, Glue
The Abbey Road reverb EQ technique not only prevents a track from swimming in reverb and thereby losing intelligibility, it also offers a means of gluing instruments and vocals together in a mix. It eliminates annoying high-frequency tails, increases clarity, yet still provides the sheen and size that reverb can impart to a sound. The trick is actually quite simple and works with any and every reverb unit; hardware or software. If you’re working with plug-ins, insert an EQ with high and low shelving filters ahead of the reverb unit. Position is important. EQ placed ahead of the reverb results in a smoother sound, since you’re equalizing the frequencies that are activating the reverb’s reflection algorithms. Since reverbs often accentuate certain frequencies, placing EQ after the reverb doesn’t have the same effect as taming the frequencies before they come in to the unit.
The essential part of this technique is to set the equalizer’s high-pass filter to a 12 or 18dB/octave slope and cut everything below 600Hz (that’s right, 600Hz). Set the low-pass filter with the same slope and cut everything above 10kHz. There you have it, the reverb EQ that Abbey Road has been using since the ’60s. However, you don’t have to stick with the high-frequency setting (the low frequency is not negotiable). You may find that cutting the highs down to 8kHz or 7kHz (or lower) works better on certain vocals or instruments in context. The idea is to reduce the highs until you can hear the vocal without the obvious high-frequency reverb tail. Another useful variation of the Abbey Road reverb EQ, in the case of vocals, is to notch out a tiny amount around 2kHz-4kHz, which will reduce possible harshness in the presence range. In the case of drums, you can cut highs further, even down to around 2.5kHz. It’s a very narrow frequency band, but it still offers all the benefits that reverb provides without adding high-frequency harshness (especially when there’s a lot of “metal-work” going on).
’Ear Now, What’s All This Then?
Unless you’re going for a special effect, such as the ’80s pre-delay followed by a high-frequency-rich tail (used most notably on percussion and snare), you can use the Abbey Road reverb technique on every instrument and vocal in your mix to achieve size, dimension, and blend. Above all, don’t be afraid to experiment. Start with the standard Abbey Road EQ settings at 600Hz and 10kHz and see where your ears take you.
In part one of this article, we discussed a way to use an affordable 8-channel Mackie mixer for “More Me” self-mixing without losing converter channels. Now we’re going to talk about a slightly more esoteric application for that 16-channel Mackie you’ve got lying around. Certain mixers, such as Michael Brauer and Andrew Scheps use a lot of parallel compression during mixing, and not just on the drum bus. Brauer, noted for his huge-sounding mixes, routes various instrument and vocal groups to one of four sub-master busses on his SSL with insert compressors. Of course, in these days of soaring track counts and artist change requests many engineers have switched to mixing inside the box, so why are we bothering to talk about routing outboard to a Mackie? Why would we not?
For those who aren’t under the gun to meet deadlines, and refuse to give up their analog ways, here’s a little trick that will enable you to route at least eight stereo outboard units to your summing mixer or interface for parallel processing without losing 16 channels—assuming you don’t have the budget to buy a pair of Rupert Neve Designs 5059 Satellite summing mixers, and a 32-channel Burl Mothership to go with them (unless you do, you lucky [email protected]$^%#).
To begin with, a little theory. You may be wondering why we’d route directly to the summing mixer and not send and return in the DAW? The reason is phase coherency. Delay compensation is never accurate in DAWs. You can hit the button to calculate delays and come back with a different number every time. Routing straight to the summing mixer keeps you in the analog realm and in phase at your 2-bus. Other reasons not to use software sends and returns to outboard include using up channels and possibly overloading converters by mixing several outboard units to two channels.
Hypothetically, since we’re dealing with a 16-channel mixer, let’s figure eight outboard units. If you have four 2-channel compressors, you’re that much closer to duplicating Brauer’s multi-bus compression. First, de-normal (output does not connect to input) the outboard to the patchbay, which means that you can send them anywhere, or in any order you wish. You’ll also need to de-normal the Mackie. Connect 16 outputs from the rear of the patchbay to the inputs of the Mackie via two DB-25 to 1/4″ TRS eight-channel snakes (or 1/4″ TRS – 1/4″ TRS depending on the patchbay connectors). Connect a third snake from the Mackie outputs to patchbay inputs (also de-normalled). You can use the main XLR outs, or the top-panel 1/4″ TRS main and Alt 3/4 outputs, which is preferable, since you can divide the outboard types by output: i.e. dynamics processors to main out, time-based and modulation effects to Alt 3/4 Out. If you have four 2-channel compressors, connect them to the eight mono input channels and pan each pair hard left and right. Set the compressors to work in dual mono (if they have that feature). You don’t want greater energy on one channel driving unnecessary compression on the other. On the other hand, reverbs and delays are fine in stereo, so if you have a Mackie with eight mono, and four stereo line input channels, connect reverb, delay, and modulation effects to the stereo channels.
If you’re using four outputs, divide the processor types between output channels. For example, compressors to main out; time-based effects to Alt 3/4. One particular advantage of this setup is that you have high-pass filters and EQ at your disposal, which means you can easily boost some high- and low-frequency EQ on your drum-bus compressor for “New York-style” parallel compression. Once you have your outboard mixed, you can either patch the Mackie outputs at the patchbay directly to the inputs of a summing mixer.
In summation (see what we did there?), along with these two outside the box uses of Mackie mixers, you can even use a 16-channel Mackie as a summing mixer. After all, that’s what mixers were designed for, and the Mackie VLZ4 series has an excellent sounding summing bus.
If you have any outside the box ideas for Mackie mixers, please feel free to share them with us.
George Adjieff, Westlake Pro CEO, sat down with long-time friend and industry pro Terry Wollman in this 5-part interview. Listen in as they discuss Terry’s rise to success as well as new and exciting projects he is working on.
Learn more about Terry on his website: http://terrywollman.com/
Photo credits in this series:
Melissa Manchester album art: Jennifer O Hill
Terry Wollman photos: Lena Ringstad
Learn more about Westlake Pro proSESSIONS events here: westlakepro.com/prosessions
Beyond the obvious use for a small-format mixer, which is, well, you know, mixing; that great-sounding budget Mackie mixer you’ve got lying around can also be a very effective studio tool for tracking and mixing in ways you may not have thought of. In this two-part article, we’ll discuss two “outside the box” suggestions for using a Mackie mixer in your project studio: A “more me” cue mixer for tracking without losing any channels on your interface, or multi-bus parallel outboard processing similar to the techniques of Michael Brauer and Andrew Scheps. All it takes is a patchbay with half-normal capabilities, some 8-channel analog snakes, and a Mackie mixer, such as the 802VLZ4, ProFX8v2, or 1604VLZ4.
If ever there was a need for an “Everything Louder Than Everything Else” button on a console, it would be for musician’s cue mixes during tracking.
“More Me” self-mixers are the current solution, but they require an input module, distribution module, and personal mixers. These systems can be very pricey, easily reaching up into the $4k price range. But there is a surprisingly affordable solution.
For this to work, you need a multi-channel interface, a TT to 1/4/” TRS, or 1/4″ TRS to 1/4″ TRS snake, and a patchbay that enables you to half-normal outputs to inputs. The beauty of half-normalling is that connecting to an output doesn’t block the input signal (“can’t stop the signal, Mal”). If you only have an eight-channel interface and no patchbay, no worries. You can pick up a Hosa MHB-350 8-point modular patchbay, which is inexpensive and comes half-normalled 1/4″ TRS to 1/4″ TRS. Plus, being modular, you can add more as you need them.
If the DAC outputs are brought to the patchbay and half-normalled to analog inputs of a summing box, outboard, monitor manager, or even open inputs, all you need to is patch the snake from the patchbay front-panel outputs directly to the line input channels of your Mackie mixer. The beauty of half-normalling is that it mults, or splits the signal, so in effect, it doubles your output channel count. Now you can send all channels to both the summing box and the Mackie “more-me” mixer.
If you’re using an 8-channel 802VLZ4, channels one and two are mono, while channel three can be mono or stereo. Since using channel 3/4 in mono reduces your channel count to seven, it’s best to use it as stereo. I would recommend the following channel assignments: vocals on channel 1, bass on channel two; drums and percussion on stereo channel 3/4; guitars on channel 5/6, and keyboards/synths on channels 7/8. If you’re using the Mackie ProFX8v2, since channels 3/4 and 5/6 can be used in mono or stereo, you have more options in terms of how many mono channels you use and which tracks to send.
Regardless, I recommend dedicating channel one for vocals, channel two for bass, and follow the same stereo output as above. Before you track vocals, if your DAW has the following capabilities, copy the bass track and pitch-correct it. Since our ear hears harmonically from bass on up, a bass that’s perfectly in tune gives the vocalist a solid pitch reference. Also, boost a little high-frequency shelving EQ on the vocal channel, which also helps the vocalist sing in tune. The beauty of the ProFX8v2 is that the singer can add comfort effects via the built-in effects processor, and you don’t have to set up aux channels.
The number of self-mixers you can use depends on how many ADA channels you have and how you allocate tracks. For example, if you have a 16-channel converter and want to use four self-mixers, put your featured instrument and its support instrument in mono, such as bass and kick drum, and the rest of the mix in stereo. With a 32-channel converter, such as an Antelope Orion 32+, using this technique, you can have up to four 8-channel mixers. It’s all a matter of creative channel allocation.
Next week we’ll talk about using a 16-channel Mackie mixer for up to eight stereo outboard processors while taking up only two input channels of a summing mixer.
Congratulations to the winners of the Butterscotch Remix Contest! Check out the announcement on Pensado’s Place:
FIRST PLACE & GRAND PRIZE WINNER – Tobias Hallhuber
SECOND PLACE – Wei-En Hsu
THIRD PLACE – Roger Dickerson