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Digging the new Vocal Rider plug-in from Waves

Just did a write up on the new Waves’ Vocal Rider plug-in for audioMIDI.com, check it out.

This entry was written by Brian, posted on November 19, 2009 at 12:13 pm, filed under News and tagged , , . Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.



Article featured in Universal Audio’s Webzine

UA LogoMy two part article on using the UAD2 system with Pro Tools has been republished in the latest edition of Universal Audio’s webzine. You can read it here.

This entry was written by Brian, posted on November 18, 2009 at 11:37 am, filed under News. 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

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 Brian, posted on November 4, 2009 at 3:06 pm, filed under Articles, PT Corner and tagged , , , . Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.



Metering with DigiRack PhaseScope

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 Brian, posted on at 2:49 pm, filed under Articles, PT Corner and tagged , , , . 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

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

  • A great starting point for most material is the default of 3 attack and 6 release, or “10 and 2 o’clock.” Set the ratio to 4:1 and adjust the input until the gain reduction (GR) starts diving a bit. Use the output control to balance out the signal, using the plug-in’s bypass as a guide. This method is great for sitting a vocal or acoustic guitar in the mix.
  • Use BF76 as a parallel processor on snare or overhead tracks (duplicate the track and run one track dry and one track with BF76). Over compress the duplicate track using a very fast release (try 6 or 7) to really bring out the sustain of the processed track then blend the over compressed duplicate track with the original to taste.
  • Try processing a mono room mic with the “all buttons in” mode.
  • Try using it on the master bus with a ratio of 4:1 or 8:1, just kissing the tops of the transients (1-3 dBs of gain reduction). You will probably have to reduce the input from the default level to achieve this.

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 Brian, posted on at 2:29 pm, filed under Articles, PT Corner and tagged , , , . Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.