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I recently mixed this record for my buddy Dominic (aka DJ Goldenchyld). It’s called Future Hop, Modern Day Renegades. Featuring Goldenchyld (of the Bangerz) on the beats and Don Prahfit on the flow. It’s cutting edge underground hip hop and it’s a free download so grab it (download here).
This entry was written by , posted on December 1, 2011 at 4:54 pm, filed under Articles, News and tagged brian's work, credits, good music. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
This entry was written by , posted on June 9, 2010 at 9:58 am, filed under Articles, News, PT Corner. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
So I’m lurking around on one of the more notorious audio message boards (you know the one, rhymes with “deer butts”) and I am checking out some of the buzz over an unreleased plug-in currently in development (I wont mention the plug-ins name because I have no gripe with the developers and actually think it could be an interesting piece of software). Anyway, the usual listening tests, commentary and e-peen measuring ensued, the trolls came out with their classic “null tests,” everything played out as expected, business as usual.
The bulk of the responses to this new plug-in were surprisingly positive, like I said, I think it could be a cool addition to a mixers toolkit but I won’t know for sure until I actually try it. But what amazed me the most was the number of posts from people fretting over the pending release date and how they would have to hold off on all their mixing projects until it came out, and we’re talking weeks not days. Site unseen, holding back mixes for something that isn’t even available to demo because of two audio samples created by the company that is selling the product sound, “pretty good.” Are you serious? I can’t believe how people continue to not get it, believing that their mixes suck solely because of their lack of some “magic bullet,” that once obtained, will allow them to transcend space-time and instantly become a better mixer; a mixer with taste, vision and a command of esthetics. By the way these posts we’re laid out you’d think Pfizer was in clinical trials with a pill for creative inspiration, “Creativia® - the viagra for your other brain.”
I could understand the first few times an amazing new plug-in or technology was announced that people might adopt this mentality, but common, the industry has been getting over on audio engineers for years with this kind of marketing bravado. It is almost like the latest plug-in is advertised and hyped like it was the latest weight loss scheme, and people are so afraid of accepting reality. Writing, recording, producing, mixing and mastering great music takes a lot of trips to the old wood shed. Practice, patience, and time to develop sensible tastes and understand the big picture. If a musician told you, “I have the most incredible song in my head, but I’m waiting for my new guitar to ship before I work it out,” you would probably laugh at them. Better yet, if an unhealthy, overweight friend told you, “I’m waiting for this new diet book to be released next month before I start trying to lose weight,” you would probably just shake your head in disbelief. Waiting to work on a mix for a plug-in you haven’t even demoed, let alone learned and integrated into your workflow is just a shame. Placing these kinds of artificial limitations on yourself will only perpetuate the idea that there is a magic bullet out there and you will end up just waiting some more when the next plug-in in announced.
Reading these posts reminded me of a story my favorite mastering engineer told me the last time I was in a session with him. He told me that he gets a call at least once a week from some amateur fader jockey claiming that the only reason they need to hire him is because they don’t have the quarter million dollars worth of gear that he has. These callers place so little value in the human element a mastering engineer adds to the point of being verbally confrontational. This made me so sad, but I totally believed him. It is not hard to believe, especially here in the silicon valley, that there are purely logic-centered people who feel that music engineering, especially mixing and mastering, is a completely objective set of skills and can be understood like any other software program or hardware schematic. This group insists that simply by having access to the tool and the operating manual will allow them to get the results they desire in a very short amount of time. It’s the “hell, if I can program this software, using it to make better music should be a piece of cake” mentality of many of these aggressively logical, tech savvy folks that tends to perpetuate this type of behavior (hint: they are generally the same types that spend more time posting on forums than actually working on music).
The reason I wanted to share these insights is not to belittle or berate this kind of mindset, but hopefully shed some light on the mistakes that I made so many times earlier in my career. You see, I was that dude who would hang on every software release, and pine for the latest audio hardware, pre-amp, or mic. I figured, shit, after programming my own modem drivers for linux, how hard can this mixing stuff be? Maybe it is a necessary part of growing into your own skin as an engineer, but a part of me wants to think that if someone politely tapped me on the shoulder and said, “hey kid, you wanna know the big secret? It’s practice, time and patience, and even then you wont know everything.” There is no doubt that great gear, both hardware and software, is a necessary part of any professionals toolkit, but at the end of the day they are just tools. Tools that provide incremental improvements to your workflow as you acquire the knowledge and experience to implement them into your work. Becoming a better mixer, producer, songwriter, or musician is an incremental and iterative process rather than a sudden paradigm shift in understanding, courtesy of some magic bullet plug-in, tip or tool. The sooner you understand this the sooner you can get on with your life and start worrying about the real magic element in art, the human element.
So my humble suggestion is, don’t wait for that plug-in, microphone, pre-amp, guitar, or *insert piece of audio gear here* to work on your art. Do yourself the favor of giving your skills and ears the benefit of the doubt. Take control of you own destiny and own up to the results, good or bad. Remember, the path to excellence in any artistic endeavor is never ending and uniquely different for everyone, don’t let a corporate road map of product releases sully the journey .
This entry was written by , posted on April 20, 2010 at 3:26 pm, filed under Articles, featured, News and tagged mixing, plug-ins, rant. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
Learning Synthesis with Vacuum
Now that Pro Tools ships with a great sounding, simple to program synthesizer plug-in, users have a wonderful way to dive into the world of subtractive synthesis. Part of the new AIR Creative Collection, Vacuum is a two-oscillator tube modeled analog synthesizer perfect for getting your feet wet in synth programming. This week at the corner I will give you a crash course in subtractive synthesis programming via Vacuum.
Subtractive Synth Basics
In its most basic form, subtractive synthesis is essentially the process of taking harmonically rich waveforms (usually generated by simple oscillators) and removing (or subtracting) some of their frequency content via filters (most commonly hi and low-pass filters). For example, by feeding a harmonically rich sawtooth wave into a lowpass filter, we can remove or reduce the higher partials to better approximate the timbre of a physical instrument (like a bowed string). By combining multiple oscillation sources with a variety of different filter and envelope options, one can generate a near infinite number of unique waveforms, some of which may emulate the waveforms of other instruments. Generally we associate subtractive synthesis with the ‘classic analog synth’ sound, a Moog Voyager is an excellent example of a subtractive synthesizer.
Synthesis in a Vacuum
While there are probably more than a hundred subtractive synth plug-ins on the market today, Vacuum strips things down to the basics with a vintage inspired, single-page interface and old-school tube sound. Vacuum is a mono-synth, much like the Moog Voyager or Little Phatty. This means you can only play one note at a time, perfect for leads and bass sounds, but you’ll want to look elsewhere for lush polyphonic pads and strings.
One of the best ways to learn any type of synthesis is to work your way backwards from a preset patch, critically examining each parameter and asking yourself how it might affect the resulting sound. While I am happy to admit that I do not have a degree in synthesis or consider myself an expert in any way, I do believe it is important for a producer to know how to ‘get sounds’ and learning the basics of synthesis key. Nine times out of ten I can find close to what I am looking for in a factory preset and with a little tweaking of a few key parameters get to the sound in my head.
Deconstructing a simple bass patch
After placing Vacuum on a mono instrument track, I have called up the ‘Bass > Woody Chap’ preset from the factory presets menu. This is a simple bass patch with straightforward oscillator, filter and envelope settings that serves as a great introduction to the concept of subtractive synthesis.
Start with the Oscillators
A synth’s sound begins with its oscillators. Remember that sound waves are made up of periodic variations in atmospheric pressure, oscillating up and down like waves in the ocean. A synthesizer’s oscillators serve as its core tone generators and will create the basic blocks of sound that we will carve our patch from.
Vacuum has two vacuum tube oscillators (labeled ‘VTO ONE’ and ‘VTO TWO’) that generate four different wave ‘shapes’ (Triangle, Noise, Saw, and Pulse Wave). The shape setting is continuous so you can create settings like 65% triangle and 35% saw. These two oscillators are combined together in the mixer to create a more complex waveform that is then fed into the filters. The rate of oscillation (and thus pitch) is determined by the note you play on your MIDI keyboard, but the octave is defined by the ‘Range’ control, with the ‘Fine’ control giving you an additional 7 semitones of pitch control in .01 semitone increments.
Notice that the Woody Chap patch uses both oscillators set to ‘SAW’ shape but separate range settings, with one an octave below the other. Play a note and change the ‘VTO1’ volume in the mixer to the right, notice the higher octave component drift in and out. At this point I could add a small amount of pitch shift to the ‘fine’ control, maybe only a 1/10 of a semitone, to achieve a subtle chorusing effect.
Move through the filters
A synth’s filters are essentially simple EQs that shape the output of the oscillator section, thus shaping the timbre of the sound. Most synths feature a low-pass filter (or LPF) with a resonance control. A low-pass filter will attenuate the hi-frequencies (beginning at the ‘cutoff’ frequency) while the resonance control will add a gain peak at the cut-off frequency. It is probably easiest to understand just by listening. Play a note and sweep the cutoff control of the LPF, increase the resonance and sweep through again. The sound you hear as a result of sweeping a resonant low-pass filter is very similar to the sound of a guitar through a wah-wah pedal. Vacuum features both high-pass (HPF) and low-pass filters (LPF). High-pass filters do the opposite of low-pass filters, attenuating the low or bass frequencies.
This patch uses no high-pass filter cutoff and a low-pass filter cutoff of 24%, aggressively restricting the higher frequency partials from the oscillator’s saw waves (remember this is a bass patch). There is a fair amount of resonance added to the LPF, so try sweeping the cutoff for a cool effect. The ‘SLOPE’ control sets the steepness of the filter and is measured in dB per octave. The envelope tracking on the LPF is positively correlated and set fairly high, meaning the filter’s cutoff will respond significantly to the envelope controls, we’ll talk about envelopes in the next section.
Enter the envelope
Most synths feature some sort of envelope that controls how the sound evolves over time, once a note is played. Think about a bowed instrument, like a violin. When bowed, the violin doesn’t immediately achieve full amplitude as it takes time before the bow causes the string to oscillate at full power. Furthermore, the tone of the instrument may change over the course of oscillation. The envelope parameters of a synth act to simulate the same concept, allowing a note to evolve over time. Vacuum features two envelope controls that by default act on or ‘modulate’ the filter and amplitude components of the instrument. The filter envelope ‘ENV ONE’ modulates the filter’s cutoff frequency while the amp envelope ‘ENV TWO’ modulates the sound’s volume.
Vacuum’s envelopes are built on the ADSR model (Attack, Decay, Sustain, Release). Each time a MIDI note is played Vacuum goes through the ADSR cycle, modulating the filter and amplitude components of the sound. ‘Attack’ defines the time it takes for modulation to reach its highest point. ‘Decay’ reflects the amount of time it takes for modulation to die down to the ‘Sustain’ level. ‘Sustain’ represents the level at which the envelope stops while the current note is held. ‘Release’ represents the time it takes for modulation to drop back to zero after the note is released. Check out the diagram for a visual representation of ADSR.
The example patch has a fairly straightforward filter envelope, where the attack (A) is set at 0ms and the decay (D) at around 80ms. What this is going to achieve is a short filter burst, moving the cutoff frequency of the LPF higher for a fraction of a second, creating a little brightness at the head of each note. To help yourself understand this, change the decay to 0ms and listen, now change it back. Notice a difference in tone? The amp envelope (ENV TWO) is set for a standard, instant-on sound with an infinite sustain. This is achieved with an attack time of 0ms and a sustain of 100%. Because sustain is 100% the decay parameter doesn’t have any effect on the amplitude. Try moving the attack time to 300ms, notice how the sound is much softer as it takes time to reach full amplitude. Set the release to 1 second and notice that the note rings out even after you have released the note. Practice understanding ADSR, knowing how to manipulate the envelopes of a patch is key to getting the sounds you want from factory presets.
Modulation Magic
Most synths allow other parameters to be modulated, outside of the envelope modulation of filter and amplitude. For example, I may want to simulate vibrato by using an additional low-frequency oscillator (LFO) to modulate the pitch of my sound generating oscillators. Many synths pride themselves on complex modulation matrices, with unlimited routing options. While this is cool for getting super tweaky, Vacuum features all the basic modulation routings you’d expect to find on a decent mono-synth. In the synth world modulation is all about source, destination and depth, or “who is modifying what and by how much.” In my example of simulating vibrato, I would make the source ‘LFO’ and the destination ‘Pitch’ using the ‘depth’ control to define the width of the vibrato. Modulation routing can be one of the tougher concepts to understand in synth programming so the best way to get a sense of it is to reverse engineer some of the factory presets.
The woody chap patch uses very little in the way of modulation routing, aside from a basic mapping of mod wheel to low-pass filter cutoff. Remember the depth controls the amount of modulation; in this case the depth controls the amount that the mod wheel opens the low-pass filter’s cutoff.
Unique to Vacuum
Beyond its basic synthesis components, Vacuum has a few unique features worth mentioning.
‘Age’ simulates the characteristics of older synths that may have unstable oscillators (drift) and worn out contacts (dirt).
‘VTA’ or vacuum tube amplifier acts as a colored master volume control. Use the shape control to add additional tube saturation to the final output. Remember, Vacuum is designed to simulate the characteristics of an analog synthesizer, so you can drive the oscillators and filters to achieve cool saturation effects. Just make sure to monitor the master volume output as to not clip the output in Pro Tools.
In Closing
Obviously this wasn’t a comprehensive tutorial on Vacuum, but more of an introduction. Hopefully I have inspired you to crack the manual or start exploring and tweaking Vacuum’s sounds on your own. A good foundation in subtractive synthesis will not only help you get closer to the sounds in your head, but also prepare you for more complex forms of synthesis down the road.
This entry was written by , posted on November 4, 2009 at 3:06 pm, filed under Articles, PT Corner and tagged pro tools, PT Corner, synthesis, virtual instruments. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
This is an excerpt from my column “The Pro Tools Corner” at audioMIDI.com
Metering with DigiRack PhaseScope
SignalTools provides a set of useful metering utilities in both TDM and RTAS formats, and come automatically installed as part of the free DigiRack plug-ins that ship with all Pro Tools systems. Consisting of SurroundScope and PhaseScope, these tools provide access to critical information regarding a signal’s level and phase coherency, both paramount in any mix workflow. This week I will walk you through the PhaseScope plug-in and hopefully shed some light on the frequently misunderstood concept of metering in Pro Tools.
Why meters matter
A nasty side effect of the ease and accessibility of DAW recording, I often find that many aspiring engineers and producers know very little about topics such as metering, headroom, gain-stages, and other basic audio engineering concepts. As modern, virtually unclip-able mixers and plug-ins become the norm in the native DAW world, many simply ignore metering all together. The truth is, meters can actually be an engineers best friend, providing vital information about a signal’s level and phase as it relates to a specific system’s output capabilities. For example, meters allow us to make sure that our signals don’t exceed the maximum level allowed by a given system or likewise, dip below the noise floor. In a digital system like Pro Tools this is extremely important, as these systems have no headroom beyond the maximum quantization level of 0dBFS, often referred to as “full scale” or “full code.” Metering is also very important in post-production and broadcast, as specific program requirements are often defined for peak and average levels. If you think about it, mixing in a system without meters would be a bit like playing a sport without the boundaries of the field marked off.
Pro Tools metering basics
The track metering in Pro Tools can be a bit convoluted depending on whether you are recording or playing back audio, and whether or not you have enabled “pre-fader metering” from the options menu. As a general rule, whenever a track is record enabled, the track’s meters display pre-fader and pre-insert input levels in dBFS, regardless of the meter option selected. This means you can rely on the track’s meters when determining the optimum recording level of a signal, regardless of the volume fader’s position and any other gain-stages added by inserts. When a track is not record enabled, the metering is governed by the option “Pre-Fader Metering” found under the Options menu. When pre-fader metering is enabled, a track’s meters display the signal level after any plug-in inserts but before the track’s volume fader is able to add or subtract any gain (hence the term “pre-fader”). With pre-fader metering enabled, a signal that came in peaking at -5dBFS with no inserts, would read the same no matter how much you push or pull the fader.
Hint: in Pro Tools HD, when a track is input enabled (but not record enabled), a track’s meter will follow the same rules as any other non-record enabled track, taking into account any plug-in inserts regardless of pre-fader metering mode.
Using the PhaseScope plug-in
The DigiRack PhaseScope plug-in is found under the multi-channel “sound field” category and can be inserted on any stereo track. The PhaseScope provides level metering with 8 different meter types (Peak, RMS, Peak+RMS, VU, BBC, Nordic, DIN, and Venue), a Lissajous meter display, and a combo phase/Leq(A) display. The combo phase/Leq(A) can be selected under the options section in the lower left hand corner of the plug-in. I generally place the PhaseScope on my master fader, as master fader inserts are the only track inserts in Pro Tools that are post-fader. In this case, by placing the PhaseScope as the last insert in the chain you are able to meter right before the signal hits the D/A at the interface, this can be useful for checking the difference in peak and/or average level a buss compressor or brick wall limiter is adding to your mix or for checking final output levels when complying with post/broadcast standards.
Setting up the level meter:
The level meter defaults to “peak” metering in dBFS, where 0 dBFS represents full scale, or the loudest signal Pro Tools can send out to the D/A converters without clipping. See the DigiRack plug-ins guide for more information on the different metering types and reference calibrations. You can set the reference mark wherever you’d like, all it does is change the color of the meter when the signal exceeds the marker (which can be very useful in post production applications where peak and average values are more scrutinized, beyond just the defacto “clipping/not clipping”). Remember, how the dBFS scale relates to the analog world is far from standardized and entirely dependent on your converter’s calibration. For example, the 192IO is factory calibrated for 18dBs of headroom at +4dBu, therefore a sinewave playing out at -18dBFS in Pro Tools would read 0 VU on an analog meter attached to the 192s +4 dBu outputs. While the complex nuances of the dB scale and all of its variations are way outside the scope of this article, if you feel up to it and want to learn more, there are some great articles just a google search away.
How to read the Lissajous and phase meters:
The goal of a phase meter is to determine how similar the left and right hand sides of a stereo signal are in relation to each other. The way the two signals relate can greatly affect the mono compatibility of a mix (as is the case where the left and right hand sides are summed into a single mono channel). While it is becoming less common for people to digest music and film on mono playback systems, phase coherency is still an important consideration in finalizing a mix. In a worse case scenario, the left and right sides of a stereo signal would be identical but have opposite polarities, resulting in a complete cancellation when summed into mono. While this rarely occurs, the phase meter can easily identify even subtle phase issues by comparing the relationship between two signals. Generally, positive values above 0 indicate acceptable mono compatibility (a value of +1 would indicate a duplicate signal in the left and right channels completely in phase), whereas values from 0 to -1 indicate potential problems.
To experiment, take two identical mono signals on two separate tracks. Pan one signal hard right and the other hard left and look at the PhaseScope plug-in on the master fader, it should read +1. Now apply the Audiosuite>Other>Invert plug-in to just one of the signals (effectively flipping its phase 180 degrees) and look at the PhaseScope again, it should now read -1. If your monitoring system allows you to sum the main output to mono, engage that now. Pretty crazy huh? Now while it is unlikely for your mix to exhibit perfect inverse phase correlation between the right and left hand sides, this extreme example can help you appreciate what is at stake.
As opposed to reading the phase meter, reading the vectorscope (or lissajous figure) in PhaseScope can take a little more practice. The goal of the graph is to visually represent the relationship between the amplitude and phase of a signal in real time. Sound complex? Well to simplify this, you can generally relate vertical lines (or lines living in the top and bottom quadrants) as in-phase, where as horizontal lines (left and right quadrants) represent out of phase material. With practice, one can even recognize different stereo recording techniques such as X/Y coincident, spaced mic, etc simply by looking at the graph.
Using the Leq(A) Meter Display
The Leq(A) display is designed to show a true weighted average of the power level in a stereo (or multichannel) signal. This meter displays a “floating” average for the level over the chosen interval (1s,2s,10s,etc). This can be very useful when trying to compare the average level vs peak level of a mix as it relates to other mixes. Experiment by comparing the average level of different mastered music in your collection (hint: try comparing something from the 70s to something from the 00s). I usually start with the default interval setting of 2 seconds. As always, remember to use your ears in addition to any metering tools, as perceived loudness can vary greatly even with two signals sharing the same average level.
This entry was written by , posted on at 2:49 pm, filed under Articles, PT Corner and tagged mixing, plug-ins, pro tools, PT Corner. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
Mixing on AIR part 4: BF76
While not technically part of the new AIR collection of plug-ins, the Bomb Factory BF76 compressor does come included with every Pro Tools installation and is the only ‘vintage’ style dynamics processor that ships with the standard configuration. Those in the know will quickly recognize the interface of BF76 as a virtual recreation of the famous Urei 1176 peak limiter, but because of its less than intuitive controls many new users forgo this little gem in favor of the more straightforward Digrack dynamics package. This week at the Pro Tools corner I will fill you in on some of the history behind the BF76 and show you how to integrate this great vintage modeled plug-in into your mixes.
Dynamics Review:
Before we can get into the specifics of BF76, it is useful to review the basics behind dynamic range and processing. The term “dynamic range” is widely used in audio engineering and is actually quite easy to understand. Simply put, a signal’s dynamic range is the difference between the softest and loudest parts of that signal’s amplitude. Dynamic range can be measured over a very short period of time, like the difference between the transient peak of a snare drum and its ringing decay, or over longer periods, like the difference between the soft and loud words of a vocal phrase. The job of a dynamics processor is to work within this realm of dynamic range by reacting to the variations in signal level that occur over time.
For example, a compressor or limiter (such as BF76) reacts to the louder portions of an audio signal by attenuating or turning the signal down by a specified amount (or ratio), with the goal of reducing the overall dynamic range of that signal over time. Think about a vocal with a wide dynamic range. Left unprocessed, many words will pop out over the rest of the mix, sounding awkward and disconnected, while others will be lost beneath the mix completely. Using a compressor or limiter the engineer is able to automatically turn down these louder words while simultaneously bringing up the softer ones, effectively reducing the overall dynamic range of the vocal. In this case, using a compressor to reduce the dynamic range allows the vocal to sit in the mix without poking out or getting lost, all without extensive volume automation.
Mixing with BF76
The bomb factory BF76 is designed as a plug-in model of the famous Urei 1176 peak limiter, developed in the late 60s by Bill Putnam and still being manufactured today by Universal Audio. The 1176 is a FET (field-effect transistor), solid state dynamics processor with a very unique sound, known for retaining the brightness and clarity of a signal that other compressors often take away. Because of its extremely fast attack and release times, the 1176 is a very versatile processor that works well on almost anything, from evening out a vocal or bass, to creating punchy snare drums, even master bus processing.
The Interface
The interface of BF76 is a near replica of its real world counterpart featuring a limited number of parameters. At first these parameters may seem counterintuitive to what you are use to seeing on other compressors, but once you learn what they do you may find that it is actually easier to use this processor.
Input: You may have noticed that the BF76 does not have a ‘threshold’ parameter like most compressors do. Many vintage compressors, including the 1176, feature a ‘fixed’ threshold that is driven by an input control, so think of the input parameter as the threshold parameter on the BF76. To increase the amount of gain reduction (or compression), turn the input counterclockwise towards 0 while watching the GR meter until the desired amount of compression is achieved. Remember, the threshold of a compressor defines the level at which the compressor begins to act on or ‘compress’ the incoming signal, in a sense it defines the “what is loud, what is not” line that the compressor uses to turn on and turn off, enabling it to control dynamic range.
Output: The output of the BF76 is used to return the signal to unity gain after compression. After achieving the desired amount of gain reduction (compression) use the output control to return the signal to its pre-processed level. It can be useful to use the plug-in’s bypass button to determine the correct output setting, as a general rule try to match the signal’s level to the bypassed state. This can help in evaluating the actual processing, avoiding the “it’s louder so it must be better” approach.
Attack: As with most compressors/limiters, ‘attack’ defines the amount of time the unit takes to grab onto the signal once a threshold breach is detected. Think of it this way, when you watch your favorite program on TV and an insanely loud commercial comes on, breaching your ears threshold of “too loud,” your “attack” would be the time it took for you to reach for your remote and turn the volume down.
The attack time on the BF76 is variable from .4 ms to 5.7 ms, which is quite fast even at its slowest setting. ‘7’ or 100% clockwise is the fastest attack while 1 or 100% counter-clockwise is the slowest attack. This is counter intuitive for most people, as one might assume that ‘7’ would be slower than ‘1’, but remember these numbers do not represent millisecond settings like most compressors, as a general rule just remember the BF76s attack/release controls are backwards. Slower attack times allow more of the signal’s transients through and can be great for putting a sharp ‘thwack’ on the head of a snare or kick drum while faster attack times can soften a signal’s attack. Because the range of attack time on the BF76 is so small, changes to this parameter can range from very subtle to inaudible, depending on the program material.
Release: Release defines the amount of time the unit takes to recover after the signal falls back below the threshold. Going back to our TV example, this would be how long it takes you to turn the television back up after the loud commercial break was over.
The release time on the BF76 is variable from 60 ms to 1.1 seconds. ‘7’ or 100% clockwise is the fastest release while ‘1’ or 100% counter-clockwise is the slowest release. Setting the release control close to ‘7’ can really help bring out the sustain of a signal for a super aggressive sound, but be careful as it can go from 0 to insane sustain and pumping with the tiniest tweak. Because the 1176 design has program dependent attack/release characteristics it is best to use your ears when setting these values rather then consuming yourself with millisecond values (notice how these values aren’t even labeled in milliseconds on the units controls). Be aware that the release control provides a much wider range than the attack and is much more sensitive to small changes, setting the attack and release times too fast can result in distortion.
Ratio: The ratio buttons define the amount of gain reduction in correlation with the threshold. For example, a ratio of 4:1 would attenuate a signal ¾ dB for every 1 dB over the threshold, so if a signal’s input is 4dB past the threshold only 1 dB will reach the output, 8dBs past the threshold at input would yield 2 dB at output. Ratios of 12:1 and 20:1 act more as limiters. Try shift clicking one of the ratio buttons to engage the famous “all buttons in” mode, dramatically changing the compressor knee and the character of the compression.
Metering: BF76 provides gain reduction (GR) and output metering modes (-18 and -24). In meter mode -18, the meter is calibrated so that -18dB FS equals 0 VU while in meter mode -24, -24 dB FS equals 0 VU. Because the less than ideal ballistics of this virtual meter don’t always help me personally, I often set it to ‘off’ and work just by ear.
Tips for using BF76
In closing
While some will argue that there are better third party plug-in recreations of the 1176 available for Pro Tools, and they might be correct, as a professional I like to be able to get the job done with just the stock plug-in set when I have to. Being familiar with all the stock Pro Tools plug-ins, including BF76, prepares you for any recording/mixing scenario, regardless of the plug-in availability on the machine.
This entry was written by , posted on at 2:29 pm, filed under Articles, PT Corner and tagged mixing, plug-ins, pro tools, PT Corner. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
Parallel Compression
Just because you failed high school trigonometry doesn’t mean you can’t use and profit from parallel compression in your mixes. Parallel or “upwards” compression is simply the process of combining an un-compressed signal with a compressed-one and blending to taste. This ‘best of both worlds’ approach is designed to preserve the dynamics, openness, punch, character and frequency response of the un-processed signal while solving the issue of the overly dynamic track getting lost in the mix or sound thin/weak.
The Setup:
While many newer dynamics plug-ins feature a built in “wet/dry” mix parameter that allows inline parallel compression tricks, you can easily achieve this effect with any compressor/limiter. Basically we are going to duplicate the track we want to compress, add a lot of compression to the duplicate and blend with the original to taste. Essentially what this is doing is creating a dense ‘bed’ of sorts for the uncompressed track to ride on, preserving all the original dynamics while allowing the track to sit comfortably in the mix. This trick can be used in subtle or extreme ways and works well on almost any source material, especially where transparent and natural sounding dynamics control is desired.
Parallel Compression Tips:
Drum Squash: Create a aux return with an aggressive compressor or limiter on its insert, you can call this “Squash Bus.” Using sends, send all your drum tracks accept the kick drum to this squash track and blend with the original (dry) drums to taste. Leaving the kick drum out prevents the squash track from over-reacting to the dynamics of the kick, which tends to dominate the other drums. You can experiment with including the kick to create cool pumping effects on the squash track. Try EQing this squash track in different ways, or even add distortion for an over-the-top effect.
Automation: Automate your parallel track up and down at different sections of the song, bring it in during the choruses for more power and support, bring it down during the verses for a more intimate feel.
Using a Limiter: For natural dynamics control on vocals, guitars, etc. try using a very fast (brickwall style) limiter on the uncompressed track just to keep the peaks under control so nothing jumps out of the speakers (you know, in that uncontrolled, “karaoke” sounding way). Then use the parallel track to subtly bring up the valleys and fill in the body/sustain of the track. This works really well on vocals with a ton of plosives (e.g. hard P or T sounds) that you don’t want to over compress. The brick wall limiter will transparently grab and control those plosives transients while the parallel track will bring up and support any softer sections without having to squash the crap out of the vocal.
Compensate for any delay to maintain phase coherency: Because some compressors/limiters incur a small amount of processing delay (usually do to look-ahead algorithms) it is important that each component of the parallel chain is delayed by the same amount. For instance, if you were to create a duplicate track of a vocal and apply a L1 Maximizer to the duplicate (parallel) track, there would be a noticeable latency and serious comb filtering would be heard when both tracks are played together. Most DAWs (except for Pro Tools) handle this automatically as part of their built in PDC (plug-in delay compensation) engine. In Pro Tools LE/Mpowered, the easiest way to solve this is to copy the plug-in to the original “dry” track and bypass it, thereby incurring the same delay on each track. A more permanent solution would be to shift the duplicate track backwards by the amount of delay the plug-in is causing. In Pro Tools HD you should always use delay compensation when mixing/editing. Remember, not all plug-ins incur a delay, in fact many do not, so considering using those when creating parallel chains.
This entry was written by , posted on October 28, 2009 at 12:10 pm, filed under Articles, MixTips and tagged compression, mixing, MixTips, pro tools. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
I swear, If I hear another R&B “Hit” with out of time vocals I am gonna go crazy. Look, autotune is fine, even appropriate in some instances. People dig it and I understand why; it’s fun, it’s a gimmick and people are suckers for gimmicks. My beef isn’t with autotune itself, it is the ~20 millisecond delay it creates when you insert it on a track. When this delay isn’t compensated for, you will hear an audible lag in the vocal track that tends to make the lyrics drag ever so slightly (not in a good way) and this definitely needs to be addressed. One would think that people making music getting played on the radio/TV would notice this delay and attempt to correct it, but somehow I continue to hear charted R&B and Hip-Hop singles with a noticeable and uncorrected “autotune” latency. Again, I’m not talking about a relaxed, “behind-the-beat” groove on the vocals, I am talking about a straight up, plug-in induced, 20-25ms lag on the entire track that is super obvious and distracting.
Why does this lag even happen in the first place? Allow me to explain. To get your busted vocal sounding like a drunk robot, autotune needs a little time to think things over and make a plan. This is where the latency, or lag comes in. Because autotune “looks ahead,” it has to delay the track a fraction of a second to do its job. Now Autotune 5 could track in real time with a lower latency than Autotune 6 (Evo) can, which has an updated pitch tracking engine, but you still wanted to correct this latency using either manual shifting or PDC (plug-in delay compensation). It is worth noting that all Antares processors with the EVO engine incur this delay and need to be compensated for.
But what if I want to sound like a robot, but correct for this latency? I’m glad you asked. Basically all you need to do is enable automatic delay compensation in your DAW (DAWs other than Pro Tools usually have this enabled by default so you don’t need to worry about it). If you are using Pro Tools LE/Mpowered (which lacks PDC), simply shift the audio on vocal track BACKWARDS in time by the amount of the delay (1380 samples to be exact). This article here explains the process in detail along with some other options. Be sure to track (record) with Autotune 5 if you want to hear the tuning while recording in real time, even if you are going to switch to Evo after the fact, the lower latency will help the performer reconcile what’s coming back through the headphones a little better. When you purchase Autotune EVO, a free license for Autotune 5 is included for this very reason, just install Autotune 5 and your ilok will take care of the rest.
This entry was written by , posted on at 12:02 pm, filed under Articles, featured, News. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
So I use quite a few “wrapped” RTAS plug-ins, meaning that the plug-in was designed for VST and hacked (via FXpansion’s VST-RTAS wrapper utility) to work as RTAS in Pro Tools. For example, all of the UAD stuff shows up as wrapped plug-ins in my inserts menu. This isn’t a big deal, as with most VST plug-ins the wrapper utility works flawlessly, but no matter which plug-in you end up using, it hides itself in the “Wrapped Plug-ins” sub-menu of your inserts menu. I have talked to Avid (Digidesign) about this and unfortunately, as long as they are “wrapped” VST plug-ins there is no way to get them into their correct categories (e.g. EQ, Dynamics, etc), aside from possibly hacking the plug-in .dpm file (I’ve looked into this. Without some sort of hint as to how the category identifier is determined/stored, I can’t see anything obvious when I pull up the files in a hex editor). While there is currently no easy way to get the wrapped plug-ins into their correct catagories, there is an easy workaround of sorts that will get you to your favorite plug-ins quickly.
Default EQ and Dynamics:
The first thing you want to do is set up your mixer’s default EQ and Dynamics plug-ins, these will show up at the highest level of the insert selection menu so put your “go to” EQ and compressor plug-ins in these two slots. To set the default EQ/Dynamics choose Setup > Preferences > Mixing Tab.
Plug-in Favorites:
The default EQ and Dynamics option only allows you to save one favorite within the EQ and Dynamics sub-menu, which blows if you want use a wrapped plug-in as your favorite EQ or compressor. In this case, you can save a plug-in as a “favorite” and it will show up at the top of the plug-ins list (within the plug-in type sub-menu on stereo tracks, e.g. TDM/RTAS or multi-channel/multi-mono).
To save a plug-in as a favorite: Hold Command (Mac) or Control (Windows) while selecting a plug-in from the inserts menu. The plug-in will not be inserted but will be stored as a favorite. Remember, you must hold Command (Mac) or Control (Windows) before clicking on the track’s insert selector.
To remove a plug-in favorite: Repeat the steps above, hold Command (Mac) or Control (Windows) and re-select the plug-in stored as a favorite from the insert selector menu.
You can have as many favorites as you’d like however keeping this list small will ultimately save you more time.
This entry was written by , posted on October 19, 2009 at 12:12 pm, filed under Articles, MixTips and tagged mixing, MixTips, plug-ins, pro tools. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
Comping with Playlists in Pro Tools 8
On top of all the fancy new UI enhancements and fantastic sounding Virtual Instruments, Pro Tools 8 has made significant improvements to many of my everyday workflows. Playlists have always been a great way to keep track of alternate takes in Pro Tools, allowing you to easily craft the perfect composite performance or “comp” by piecing together different takes. Pro Tools 8 has enhanced this process infinitely by allowing you to view, edit and audition a track’s playlists within its new “Playlists View.” This week at the corner I will show you some tricks that will cut your comping time in half and share what’s new with playlists in Pro Tools 8.
Playlist Basics
Essentially playlists are just a way for Pro Tools to store the timeline placement of a group of regions on a track. Regions are pointers to raw audio files on your hard disk, while a playlist stores the organization or placement of multiple regions in time. A track in pro tools can have an unlimited number of playlists associated with it, or “virtually unlimited” as we like to say in the DAW world. In other words, you don’t need to worry about running out.
New loop record workflow
Before Pro Tools 8, loop record and playlist comping were sort of mutually exclusive workflows. You generally used either the “takes list” comping method with loop record, or you stopped recording and manually created a new playlist after each take. Fortunately, these two features now work harmoniously in version 8, using the new preference “Automatically create new playlists when loop recording.” When this option is checked, Pro Tools automatically appends each new pass in loop record to a fresh playlist. Let’s check it out.
Step 1: I start by checking the preference “Automatically create new playlists when loop recording,” found under the Operations tab of Setup > Preferences.
Step 2: After creating and naming a new track for my loop record pass, I enable loop recording via Operations > Loop Record or by using the shortcut Option-L (Mac) Alt-L (PC).
Step 3: At this point, I need to make a selection defining the “loop” that I will record each pass over. Remember, loop record mode always requires a selection to define the length of each take. If you need a refresher on the basics of loop record mode, check out my previous article here [http://www.audiomidi.com/classroom/protools_corner/ptcorner_67.cfm].
Step 4: After defining my loop record selection, I can add a bit of pre-roll to get me into the first pass and record enable my track. To make this easier, Pro Tools 8 added three new shortcuts for record, solo and mute track. Shift-R for record, Shift-S for solo, and Shift-M for mute on the selected track.
Step 5: In this example I have recorded 3 takes and Pro Tools has created 3 new playlists, leaving the final pass as the active, or “main” playlist on my track. Because Pro Tools left my final take on the original playlist “LoopRec,” I will double click the track’s nameplate and rename this playlist to “LoopRec.03.”
Tip: You don’t have to complete your loop recording in the initial run, you can stop and start up again as much as you’d like. Because Pro Tools only creates a new playlist for the second pass of a loop record take, just create a fresh playlist for each loop record set you want to do.
Tip: If you have already completed a loop record pass without the preference “automatically create new playlists in loop record mode” checked, you can simply right-click one of the loop recorded regions and choose Matches > Expand Alternates to new Playlists. Alternates are defined by the “match criteria,” to change the match criteria right-click a region and choose Matches > Match Criteria.
Comping with the playlists view:
Now that I have a set of playlists, either ones created automatically via loop record or ones created manually during the recording process, I can easily view these simultaneously by switching the track’s view to “Playlists.” Click on the word “Waveform” and select “Playlists,” or Ctrl+Opt+Cmd-Click (Mac) Start+Alt+Ctrl-Click (PC) on the track’s playlist selector.
Right now the active or “main” playlist is LoopRec.01, but I want to create a new main playlist for my composite take. I can do this by clicking on the track’s playlist selector (the little down arrow next to the track’s name) and choosing “New…” I name this playlist LoopRec.Comp and since it was the last playlist created it now becomes this track’s active or “main” playlist. Remember, whichever playlist is active (selected via the track’s playlist selector) is by default the track’s main playlist, therefore any playlist can be the “main playlist” for a given track.
Hint: When selecting playlists, you can generally ignore the number in parenthesis after the playlist’s name (e.g. “Playlist.01 (XX)”). This is simply a master playlist counter telling me when the playlist was created relative to others. This continues to count up even after tracks/playlists have been deleted or after an undo.
To audition the main playlist for a track I simply hit play. To audition alternate playlists associated with a track, I simply hit the solo button (the “S”) on each alternate playlist. When auditioning, try making a selection and using the key commands Cntrl-P/Cntrl-; (Mac) or Start-P/Start-; (PC) to move the selection up and down in conjunction with Shift-S (track solo key command) to quickly audition each alternate playlist without using the mouse. Tip: In command key focus mode you can use ‘P’ and ‘;’ without a modifier to move the selection up or down.
To promote a selection to the main playlist, simply select the piece you wish to copy, right-click and choose “copy to main playlist,” or use the key command Cntrl+Opt-V (Mac) Start+Alt-V (PC). You can also copy the selection to a new or duplicate playlist from the same menu.
Once you are finished editing you can re-hide the alternate playlists by switching the main track back to “Waveform” view. The alternate playlists will still remain in the session, associated with the track in case you need to do any further comping.
Remember, all of these playlist and loop record workflows will work with MIDI data too!
Rating Regions:
Pro Tools 8 features a brand new region rating system that allows you to give any regions a numerical rating of 1-5. Use this to rate each pass in a loop record take, and then use the playlist view’s “filter lanes” function to show only takes with a rating of 4 or better.
To rate a region: Simply right-click on any region in the edit window and choose Rating > 1-5.
To display the region’s rating: Choose View > Region > Rating.
To filter the playlist lanes: In playlists view, right click on any playlists name and choose “Filter Lanes.”
Tip: each region has a its own numeric rating, if you have already edited a group of regions and wish to rate them as a whole, first consolidate the regions into a new whole region using Edit > Consolidate Region (currently region groups and the rating system don’t work so well together)
Some considerations
While loop recording with new playlists is a great way to speed up your recording and comping workflow, there are a few things you may want to consider. For example, let’s say you loop record 3 passes of your first verse and then you want to loop record 3 passes of your second verse. The system will create an entirely new set of playlists for the second verse’s loop record pass, leaving you with 6 playlists (3 representing the first verse’s takes and 3 representing the second verse’s takes). So depending on how you are used to using playlists and loop record for tracking and comping across a complex multi-part tune, just work out a organizational game plan in your head before hand, otherwise you may end up with 40-50 playlists on each track (which is fine if that is what you want). This is a situation where region rating and lane filtering can really be handy. Imagine a track representing 3 verses of a song. Each verse has 10 takes generated via loop record for a total of 30 playlists. After auditioning and rating each take, the lane’s filter can pair your choices down to only the 4 and 5 star takes, or filter based on regions within the timeline selection, making the comping process much more organized and efficient.
This entry was written by , posted on October 18, 2009 at 7:22 pm, filed under Articles, PT Corner and tagged editing, playlists, pro tools, PT Corner. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
