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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.
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.
This is an excerpt from my column “The Pro Tools Corner” at audioMIDI.com
Pro Tools 8: MIDI Editor
The MIDI editor in Pro Tools 8 is one new feature that has certainly been turning heads, especially those of other “sequencer heavy” DAWs. While many users have been making comparisons to Logic’s flavor of MIDI editing, the Pro Tools MIDI editor features a few unique tricks that are sure to speed up your sequencing workflow. This week at the Pro Tools Corner I will walk you through the basics of the MIDI editor and help you unlock to true sequencing power of Pro Tools 8.
Opening the MIDI Editor
First off, if you are worried that your current MIDI workflow isn’t going to translate in Pro Tools 8, fear not. You can still work with MIDI data in the edit window just as you had in Pro Tools 7 and earlier, but once you experience the MIDI editor you probably won’t want to.
Working with the MIDI editor is fairly straight forward, and the editor can be accessed in one of two ways:
To open a floating MIDI editor: With the MIDI or Instrument track view set to “regions,” double-click with the grabber tool on any MIDI region or simply select a group of notes and choose Window > MIDI Editor.
Note: double-clicking with the grabber tool in PT 7 and earlier brought up the “rename region” dialog, you can return this functionality in the MIDI tab of your preferences.
To open a “docked” MIDI editor: Choose View > Other Displays > MIDI Editor. This will dock a MIDI editor at the bottom of the Edit window.
If the windows “target” button is active (red), this docked editor will update dynamically as you select new MIDI data in the Edit window and can be resized vertically to the size of your choice. The “target” is the small square in the upper right-hand corner of a floating window.
Hint: Save a window configuration of the docked MIDI editor to toggle it instantly.
Using the MIDI Editor
The beauty of the MIDI editor is that its toolset, zoom settings, grid resolution and edit modes are completely isolated from the Edit window. For example, you could be using the smart tool in grid mode with a resolution of 1 bar in your Edit window and have the pencil tool in slip mode active in the MIDI editor. Generally, all of the tools in the MIDI editor will work the same way as they do while editing MIDI in the Edit window, so you don’t really need to learn any new edit tools, you just have get used to looking at MIDI in another window.
Note: If you are used to using single key shortcuts with command-key focus, you will need to focus them to the MIDI editor while docked. Command-key focus is represented by the “a-z” button in the top right hand corner of the editor.
Like the Edit window, The MIDI editor allows you to view additional MIDI/Instrument track data below the piano roll. By default, Velocity is shown but additional lanes can be shown or hidden by clicking the “plus” or “minus” icons. This can be very handy when you are editing multiple layers of CC data.
To quickly switch the editor into score view, simply click on the score button at the top left hand corner of the editor.
Viewing MIDI in Layers
By default, the MIDI editor shows only the selected track’s MIDI data. By using the track show/hide list attached to the MIDI editor, you can actually view MIDI data in overlapping layers. Think about editing multiple tracks of MIDI drums within a single piano roll and you will understand the power of this special feature.
The black dot to the left of the track name represents the track’s show/hide status, while the pencil to the right lets you know which track you are currently editing.
When you work with MIDI in layers, by default each layer will be represented by its region color. Because many times you may have regions that are the same color, the MIDI editor features two alternate color-coding options for notes.
Color notes by track:
Regardless of track color or region color, this option will assign a different color to each note layer in the MIDI editor. This helps when two tracks have the same region color.
Color notes by velocity:
This option will color code notes based on their velocity. Darker colors represent higher velocities. You can set all MIDI notes to color code by velocity automatically within system preferences, under the “Display” tab.
MIDI Editor Options:
You can set custom scrolling options in the MIDI editor using the discloser triangle at the top right hand corner of the editor
Just like the edit window, you can view additional ruler displays (like meter, tempo, etc).
In Closing:
While it might take you a little time to adjust to editing MIDI in a different window, I think you will find that the new editor makes working with MIDI in Pro Tools much more efficient and intuitive. In the past, I would always have trouble getting the correct level of horizontal, vertical zoom and octave range on my MIDI tracks, the MIDI editor has completely eliminated these navigational inefficiencies for me, and with a little practice I think it will for you as well.
This entry was written by , posted on at 6:28 pm, filed under Articles, PT Corner and tagged editing, MIDI, 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.
Pro Tools Automation Part 2:
In the last installment of the Pro Tools corner, I walked you through some basic automation techniques in Pro Tools. Picking up where we left off, this week I will show you a few more ways you can manipulate automation in Pro Tools, including graphic editing and the automation of plug-in parameters. If you missed the first part of this article, you might want to check it out here before you proceed.
Graphic Manipulation of Automation
In the last article I showed you how to view your automation data directly against the track’s waveform and hinted at the fact that the automation points, called “breakpoint” can be manipulated with the edit tools. Some people like to call this “graphic automation” and it is a great alternative or supplement to the real-time automation we learned last week, especially if you don’t own a control surface.
Remember you can switch a track’s view to show any automation graph by simply clicking on a track’s view selector.
Editing Automation with the Grabber Tool:
Editing Automation with the Pencil Tool:
Editing Automation with the Trim Tool:
The trim tool is used to scale existing breakpoints up or down, or make “delta” changes in an already existing automation graph. The trim tool works best by first selecting a range of breakpoint with the selector tool and then trimming them up or down. Notice the “tooltip” in the top left-hand corner of the graph that shows you the current parameter value as well as the “delta” amount or relative change. The trim tool is great for selecting a short passage, say a phrase or word of a vocal, and easily trimming it up or down by a few dBs. Like the grabber, you can hold down the Command key (Mac) or Control key (PC) while you trim for finer trim increments.
Other Editing Tips:
Automating Plug-Ins
Automating plug-ins is pretty much identical to automating volume or pan in the mixer, but there is just one extra step before you start. Because complex plug-ins can have hundreds of automate-able parameters, it would be inefficient to have all these show up under a tracks automation view when most of the time you are only interested in automating a few specific parameters. Because of this, Pro Tools requires you to enable the specific plug-in parameter you wish to automate before adding automation.
To enable a Plug-ins parameter for automation:
Alternatively you can look at the plug-ins entire list of automate-able controls by clicking on the plug-ins automation button, located underneath bypass. This list contains a left hand side of potentially automate-able parameters and a right hand side of parameters currently enabled for automation. Double click a parameter to move it from the left list to the right list. Pro Tools can by default set all plug-in parameters to be enabled for automation as soon as the plug-in is inserted. The preference “Plug-In controls default to auto-enabled” can be found under the Mixing tab of Setup > Preferences.
This entry was written by , posted on at 1:38 pm, filed under Articles, PT Corner and tagged automation, mixing, 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.
The following is an excerpt from my column “The Pro Tools Corner” at audioMIDI.com. It was written for Pro Tools 7 but basic automation workflows havn’t change.
Pro Tools Automation 101
One of the most useful feature sets found in nearly all of today’s DAWs is automation. Traditionally reserved for only the most expensive large format consoles, automation allows you to record parameter changes in the session’s mixer and is a must know technique for any serious Pro Tools user. This week at the Pro Tools Corner, I will walk you through some of Pro Tools’ basic automation features and show you how to record and edit automation in your session.
Why we automate:
While there are many reasons to use automation in Pro Tools, both creative and otherwise, on a basic level automation lets us take once static parameters in the mixer and allows them to change dynamically over the course of a session’s timeline. For example, one may find that while a specific volume level on a lead vocal works well during the verse the same level doesn’t work as well during the chorus. Without splitting the vocal out onto separate tracks, automation provides an easy solution to the level changes needed over the course of the song. Ask around, most of the world’s top mixers would certainly agree that effective use of automation is a huge component in achieving a great sounding, dynamic mix. Pro Tools takes basic level automation a bit further by allowing the user to automate volume, pan, mutes, send levels/pans/mutes and plug-in parameters. Users can record fader moves and parameter changes during playback in real time or edit automation graphically offline.
Recording automation in Pro Tools
Just a like a fancy SSL or Neve console with flying faders, Pro Tools supports the recording and playback of real time fader moves and parameter changes in the mixer. To record changes in the mixer’s state over time, one can simply select a real time automation mode from the track’s automation mode selector and playback the session. Pro Tools will then record any changes made in the mixer during playback and depending on the automation mode selected, will update or ignore existing automation data.
To record real time level/pan/mute/send automation on a track:
Ensure that automation is not suspended and that the parameters you wish automate are write enabled (highlighted in red) in your session’s automation window (Windows > Automation). Any parameter that is not “write enabled” will be ignored when recording real time automation.
Click the track’s automation mode selector (this can be found in both the edit and mix window) and switch it from “read” to “write” in the drop down list. If you want to automate more than one track at a time, simply set those tracks automation to “write” also.
Now playback the session and move the faders, pan, etc as desired. You do not need to record, just playback the session. Automation has its own “record enable” as outlined in step 2 and recording automation has nothing to do with a track’s record enable or the transport master record status.
After an initial automation pass, the track’s automation status will change from write to touch or latch (this is actually a preference found in the mixing tab of Windows>Preferences). This will prevent you from accidentally recording over any previous automation when playing back your track.
Now playback the session and watch those faders fly!When you are finished automating a particular track, set its automation mode back to “read.”
Note: While it is nice to perform in automation with a control surface, you can also use your mouse to record automation in real time.
A quick note on plug-ins:
While plug-in parameters can be automated almost as easily as volume/pans/mutes, it does involve an extra step, which will be coved in part 2 of this article.
Pro Tools automation modes:
All Pro Tools systems feature the following track automation modes:
Off: Automation is suspended for that specific track, mixer parameters revert back to manual control. To suspend all automation in a session, use the master suspend found in the automation window (Windows>Automation)
Read: Previously recorded Automation is played back (if it exists).
Touch: Pro Tools only records automation when a parameter is modified or “touched” but acts like read mode otherwise. This is commonly reffered to as an “update” mode, allowing you to update a previous write pass, appending new automation data only where desired.
Latch: Similar to touch, latch acts like read mode until a parameter is modified or “touched.” After being modified the parameter does not return to its previous state and remains “latched” into its current position.
Write: A destructive mode, write will record any incoming automation disregarding previous automation in the track. Write will destructively “burn over” any and all automation on a track, meaning even if you don’t touch a parameter its current state is being recorded during the entire pass.
Pro Tools HD systems also feature Touch/Latch and Trim automation modes, which we’ll save for another article.
Viewing your automation:
One of the greatest things about working in the computer is the ability to actually see the automation we have recorded, represented visually as a set of breakpoints. By selecting the specific automation graph from a track’s view selector we can not only view the existing breakpoints, but also manipulate them graphically using our edit tools.
To view a tracks volume automation graph:
Switch the track’s view selector from “Waveform” to “Volume.”
Automation graphs are a series of breakpoints connected by a solid black line, sort of like a game of connect the dots. The mixer reads these break points like vector data to change parameters smoothly over the timeline from one breakpoint to the next.
You can select other automation graphs from the tracks view selector, including volume, pan, mute, and send level/mute views.
Note: Automation graphs for sends show up as sends are assigned in the mixer, i.e. if a track doesn’t have any sends, you wont see any automation views for them in the track view selector.
Remember, automation in Pro Tools lives on the track and each track has only one set of automate-able parameters (ie: switching playlists does not switch a tracks automation graph). Once a track contains even one automation breakpoint, it will no longer respond to manual control (unless the track’s automation mode is switched to off, or automation is suspended). To get around this, simply insert the “Trim” plug-in on your track and use that for track wide, or “delta” changes in volume level.
Coming up in part 2:
Stay tuned, in the next installment I will walk you through graphic manipulation of automation breakpoints and show you how to automate plug-in parameters.
This entry was written by , posted on at 1:37 pm, filed under Articles, PT Corner and tagged automation, mixing, 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.
This article was written back when 7.4 came out (introducing elastic audio), but is still equally relevant in Pro Tools 8 as these features haven’t changed much.
Elastic Audio Part 2:
In the last installment of the Pro Tools corner we took a first look at 7.4’s incredible new elastic audio feature. This week I want to dive a little deeper into the topic, showing you how to easily quantize and warp audio directly in the timeline with incredible accuracy and sound quality.
Quantizing audio regions elastically:
The beauty of elastic, “tick” based audio is its ability to re-conform itself much like MIDI data, making tempo changes and quantization (or re-grooving) a breeze. What use to take hours of laborious editing time can now be done in a few clicks, and still sound better then doing it the old fashioned way with beat detective.
To quantize an audio region:
Manually warping audio events:
Sometimes, rather then quantizing an entire selection of audio events, all you want to do is manually shift an event slice one way or another. In this case, Pro Tools allows you to easily create and shift warp points within a region manually. For example, you may want to carefully manipulate the timing or feel of a vocal track, a common situation where an outright quantize would yield less then satisfactory results.
To editing a regions warp points:
If you ever need to remove the warp properties from a region and revert back to the original copy, simply right-click on the region and choose “remove warp.”
Correcting Transient Analysis:
The secret behind Pro Tools ability to stretch, shrink and quantize audio regions accurately lies in its transient detection, or the ability to identify rhythmically significant points within a file. If Pro Tools is detecting too many erroneous transient points, you can edit a regions “event sensitivity” via the region’s “elastic properties.” Adjusting the detection sensitivity is similar to using the sensitivity slider in beat detective, lowering the sensitivity on material with complex waveforms and poor transient definition can help remove excess analysis points (ie: vocals, complex synth patches, etc).
To edit a region’s elastic properties right click on the region and choose “elastic properties.”
In the rare case that a region’s transients are not properly identified and adjusting the event sensitivity doesn’t help, you can manually edit a region’s analysis using the grabber tool inside the track analysis view.
To manually correct or modify analysis:
Because editing analysis points on complex polyphonic material can be quite tricky, start by practicing on rhythmic material with clearly defined transients.
This entry was written by , posted on October 14, 2009 at 6:21 pm, filed under Articles, PT Corner and tagged editing, elastic audio, 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.
This article was written back when 7.4 came out (introducing elastic audio), but is still equally relevant in Pro Tools 8 as these features haven’t changed much.
Introducing elastic audio:
A few weeks back we took a first look at some of the new features available in Pro Tools 7.4, the most notable being elastic audio, or the ability to stretch and squeeze audio regions automatically in the timeline. After spending some quality time with the new features, I must say that elastic audio is simply amazing and will definitely transform the way you work in Pro Tools and save you hours of time. While there are surely an endless number of uses for elastic audio, this week I want to walk you through a basic audio loop workflow inside Pro Tools 7.4.
Elastic Loops
One of the greatest features added in Pro Tools 7 was the ability to drag and drop audio content from the finder or windows explorer directly into your session, allowing you to quickly add loops and audio samples from anywhere on your hard drive. While this was nice, the usefulness of audio loops has always been limited to the source material’s tempo, unless you want to spend the time using beat detective or manually conforming the audio material to your session’s tempo. Now, by taking advantage of 7.4’s elastic audio engine, not only can we drag and drop our favorite loops directly into a session, but these loops will also re-conform themselves automatically to the session’s tempo map.
If you have ever used apple loops in logic or garage band, or worked with Abelton live, you may already be familiar with the benefits of an elastic timeline. The workflow that I am about to show you will help get you started with elastic audio in Pro Tools, as well as showcase some of the unique elastic time features inside 7.4. Remember, to complete this example you will need to have Pro Tools 7.4 installed on your system, as the techniques I am about go over will not work in previous versions.
Example: basic elastic audio workflow in 7.4
I have prepared for this example by first creating a new blank session into which I will import and time my audio loops. I have chosen a few different loops to work with in this example (a simple drum, bass, and percussion loop), but almost anything will work. As you start to experiment with elastic audio, you may want to try working with rhythmic tracks first, as their transient detection confidence is very high and they can be re-timed relatively easily.
Step 1: Setting up
Before I begin, I want to check a specific option location in the processing tab of Setup > Preferences called “enable elastic audio on new audio tracks.”
Step 2: Importing the first loop
To import my first loop (a drum loop) into my session, I can simply drag and drop the file from my finder (or window’s explorer) into the tracks list in Pro Tools. A new track will automatically be created, and since this is the first file to be imported Pro Tools will ask me if I want to import the tempo from the loop.
In this example I will go ahead and choose “import” from the dialog. Pro Tools will now adjust the tempo ruler to match the extracted tempo from the audio loop.
Step 3: Selecting tick based audio
Because audio is generally sample based (), Pro Tools defaults the track’s time base to samples. When using elastic audio in conjunction with tempo changes, I want to change the track’s time base to “ticks” (
) by clicking on the track’s time base selector.
Now I can change the session’s tempo and all tick-based tracks will follow the tempo ruler just like MIDI events.
Step 4: Previewing new loops
Now that I have the tempo extracted from the first loop, I want to preview some new loops at the current session tempo, before adding them into the session. To do this I will browse and import my loops from the workspace browser, located under Windows > Workspace.
To preview the loops at my current session’s tempo I will click the “audio files conform to session tempo” icon in the workspace browser.
To preview a loop in the workspace browser, simply select the file and hit space bar or click on the preview icon. Note: you can choose to preview with loop playback or activate auto-preview by checking those options in the browser menu.
To import a file from the browser I can again just drag and drop it into the tracks list. A new track is created and automatically set to ticks with elastic audio enabled. I can continue to preview loops and add them into my session all at the same tempo, regardless of the loop’s original tempo.
Notice the elastic audio icon in the top right hand corner of the region, and next to the region name in the region’s list.
Step 5: Changing the tempo
Now that I have imported a few loops and set them up to re-conform as elastic audio, I can easily change the manual tempo in the transport or even add tempo events in the tempo editor. Notice how the regions squeeze and expand to match the bar|beat grid.
Quality considerations:
Now is probably a good time to discuss some of the fidelity considerations when stretching or shrinking audio files. While the elastic audio algorithms in Pro Tools are very good, as a general rule of thumb you don’t want to stretch or shrink your audio too much (your ears will tell you how far you can go). You can help the process out a bit by selecting the appropriate plug-in algorithm from the track’s elastic audio drop down menu. Pro Tools features 5 different base algorithms to choose from: Polyphonic, Rhythmic, Monophonic, Varispeed, and X-Form (rendered only). While a complete breakdown on the differences between the algorithms is beyond the scope of this article, try experimenting. Start with the appropriate algorithm for the type of audio you are working with and then try out different presets, listening for any changes.
More elastic audio coming up:
We have only begun to scratch the surface of elastic audio in Pro Tools. Stay tuned for elastic audio: part 2 with more tips, tricks, and tutorials on this amazing new feature.
This entry was written by , posted on at 6:00 pm, filed under Articles, PT Corner and tagged editing, elastic audio, 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.
This is an excerpt from my column, “The Pro Tools Corner” at audioMIDI.com, it is pretty old (2008) so I have since updated it to reflect newer version of Pro Tools hardware and software.
Updated 2012: Pro Tools 10 introduced HDX which uses a 64bit floating point mix bus and 32bit insert chains on it’s DSP based AAX plug-ins, effectively making it sonically identical to the Pro Tools Native system mixbus, RTAS and AAX native plug-ins. Older Pro Tools HD TDM (accell and process) cards will continue to use the 48 bit fixed mixer and 24-bit insert points until they are not supported in Pro Tools 11.
Update 2: As of Pro Tools 9 it was announced that Pro Tools Native uses a 64 bit floating point mix bus for internal summing and 32 bit float insert points, this was only revealed when the Pro Tools HD native card was released. Prior to this, it was stated that Pro Tools LE had a 32 bit floating point summing mixer, but it has come to my attention that it has been 64 bits for quite some time now (even before Pro Tools 8), although this was never revealed most likely due to marketing concerns over a perceived difference between the 48-bit TDM mixer found in the more expensive legacy HD systems.
Master Faders Demystified: Part 1
Often confused with a generic “volume control,” the master fader is one of Pro Tools most confusing, yet supremely critical features. Is it a master volume? Does it add an additional gain stage like an auxiliary track? Why don’t they have “inputs”? By bridging the extended headroom capabilities of the mixer with the 24 bit reality of your D/A converters, the master fader serves as a final output trim, allowing the mixer to optimize the signal coming off the high resolution internal mix-bus as it exits the system into the interface’s converters and ultimately the analog domain. In this two part series, we will take a look at the mysterious master fader and I will attempt to demystify its usage in Pro Tools LE/M-powered as well as Pro Tools HD.
Understanding mixer headroom
Before we can get into the nuts and bolts of master faders, it is important to review the concept of “headroom” and how it relates to the Pro Tools mixer. Generally, the term headroom is defined as “the dynamic range between the normal operating level and the maximum output level or clip point.” In an analog system, this concept is a bit grey in practice, as a good analog mixing console has a boat-load of headroom and can even sound subjectively better when pushed passed the “clip” point. In a digital system however, things are a bit more black and white. Measured in dBFS, a digital system’s clip point is hard at 0 dBFS. Meaning any signal that exceeds a value of 0dBFS is gone. Not subjectively saturated, or smoothly compressed like the heralded non-linearities analog tape or tube gear, but hard clipped in that nasty, “digital distortion” kind of way.
So how exactly does this relate to the digital recoding process and the pro tools mixer? Well, when we record audio into Pro Tools, we generally do so at a resolution of 24 bits. Meaning that each sample of audio we capture uses a 24 bit word, or 24 1s and 0s to describe the signal’s amplitude at that point in time. Likewise, in a 24 bit digital system, we can’t measure amplitude louder than what is represented by a full code sample (all 1s). We hear the ramifications of this all the time when we record too hot into the system; the input signal’s amplitude exceeds the maximum amplitude that can be described by the 24bit system and the loudest part of the waveform is clipped off, or thrown away at the converter, yielding that nasty digital distortion. But clipping doesn’t only occur at the recording stage, it can also occur when lots of very loud signals are summed together inside a digital mix bus.
Imagine a 4 bit recording system, if you were to take two words at full code and combine them together: “1111” + “1111”, you would need a 5th headroom bit to describe the resulting output of “11110.” Or think of it this way, if you had 2 one-gallon buckets and they were each nearly full of water, it would make sense that you would need a larger bucket to combine the two smaller buckets into one, without spilling any water. The same is true for a mix bus. If we want to add many “hot,” or near full-code 24 bit signals together, we would need a mixer with a higher precision (say 32 bits, 48 or 64 bits), or more “headroom,” for them to sum nicely without having to turn each down individually as it enters the mix bus. This is exactly what Pro Tools provides for us. In Pro Tools LE/M-powered and Pro Tools 9/10 Native, the mixer sums signals together using 64-bit floating point math, using 56 fixed bits (called the mantissa) with 8 additional exponent bits (think scientific notation) to scale and maintain precision as signals are added across it. Without getting into the complexities of floating point arithmetic, it is safe to say that the Pro Tools Native mixer has a ton of head room, so much that you are not likely to clip the internal mix bus, even if you had 256 24 bit tracks of full code audio, with each fader at +12dB running through it. The Pro Tools HD TDM mixer is a different story, we’ll get to that later.
Enter the master fader
At this point you must being thinking, “that’s BS, I have totally clipped the pro tools mixer. I see clip lights all the time and I hear distortion at my output.” Technically speaking, what you have clipped is your converter, which does have a maximum output of 24 bits (clipping at 0 dBFS), and is therefore fairly easy to clip with even a few hot signals in the mix. You can also clip signals during recording, as your ADC can only “see” or measure a maximum of 24 bits of signal on input, and occasionally you can clip plug-ins (as they have their own internal calculations going on). This is exactly where master faders come in. Master faders allow you to meter the summation of all the tracks in a mix as well as trim the output before the signal is truncated and exits the system to the converters at 24 bits again. Think of it as a “bit selector” of sorts. If you had a system with an internal resolution of 48 bits and the output could only “see” 24 of those, the master fader effectively puts a handle on which output bits you choose to leave the system with. If you didn’t have that handle, you would have to individually reduce the level of each track as it enters the mixer to avoid clipping the converter. On top of this additional headroom, the mixer also allows for “footroom” bits to preserve the resolution of lower level signals that were attenuated significantly by the mixers level control. In other words, you need not be concerned about “eating into” a signals bit depth by reducing the volume of an individual track in the pro tools mixer.
How to use the master fader as an output control
In Pro Tools Native, LE or M-Powered, implementing the master fader is a fairly painless procedure. Simply create a new stereo master fader (Tracks > New) and use it to monitor the overall output of your mix. It should automatically set itself to control your main outputs (generally A1-2) and If the clip indicator lights up, it’s time to pull down the master fader until you are leaving the system with a nice, un-clipped output. Likewise, if the summation of many tracks seems to weak, you can kick up the master fader to optimize the output. Contrary to many rumors, the master fader doesn’t add any sonic color to your mix, nor does leaving it at unity gain or not using one in your session benefit you in any way.
At no time should you use the master fader as a “volume control” for your monitors, that is what your Mbox 2’s, 003’s, (or whatever PT interface you use) monitor or headphone level control is for. Your speakers/headphones are analog devices and their input level should be controlled in the analog domain, not the digital one.
Exercise: Sine Wave Test
To demonstrate the master fader’s purpose, try the following:
What is happening? The combined input of all those sine waves into the mix bus is clipping the output at the converter but not the internal mix bus, the master fader allows us to recover and exit the system with a clean tone. Even if we were to submix some of the tracks into a bus and back into a new aux track, the signal master fader at output has us covered in the native world of (Pro Tools HD TDM is a totally different set up, stayed tuned).
So why do individual track clip lights matter? When recording into the system these help us avoid clipping the converter’s input and are critical in determining the input trim of your preamps. Once a signal has already been captured, during mixing, they really don’t mean anything, as long as you manage your final output (again this is different in the 48-bit HD TDM mixer). However, they do matter If you are mixing to a summing mixer, especially if you are setting each track’s output in the mixer to a discrete interface output. Remember, you can clip the converter at input and output, but it is highly unlikely that you will clip the internal mix bus in Pro Tools.
Coming up next…
In Pro Tools HD TDM (Not Pro Tools HD Native, that uses the 64 bit floating point mixer) the story is a bit different, because the mixer runs at a fixed 48-bit precision and is truncated back to 24 bits at each insert or input point in the mixer, headroom can be a little more challenging to manage. Next time we will look at good practices for master faders in Pro Tools HD TDM and cover some tips for using Master Fader inserts with “mastering” style effects and dither in your mixes.
This entry was written by , posted on October 13, 2009 at 1:12 pm, filed under Articles, PT Corner and tagged mixing, 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.
