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So I’m not sure how I missed these (probably because I am off in Pro Tools land and forget to check out all the amazing developments happening in the VST and AU plug-in world) but I just recently checked out the plug-ins from Stillwell Audio and they are so beyond legit. I am really digging the “Event Horizon” clipper/limiter, it’s freaking incredible. Most brick wall limiters let you get things loud, which is nice, but they tend to pump, jack the imaging and add a lot of color to the signal and sometimes that’s not cool. Mastering engineers will often forgo a brick wall limiter in favor of just clipping the front end of some really hi-end A/D converters (obviously you must do this very carefully and you can definitely tell when you’ve pushed to far). Anyways, Event Horizon kind of emulates this practice in the box, and while you can’t use it like a L2 style brickwall limiter, and you definitely have to know when to stop, it sounds so much better than your standard “look-ahead” style limiter on mixes where maximum RMS is not required (read: mixes that don’t need to be ridiculously loud for “competitive” reasons). Event horizon is also pretty sweet on drum tracks, you can push your kick and snare into it just enough to get the transients to settle into the mix while not eating them alive like most limiters. Recently I was mixing a sort of “Beatles meets Beck” style tune and this really captured the drum sound I was after, a little distorted and ruckus but not like running your drum bus through sans amp kind of ruckus.
Sadly, the Stillwell stuff is currently only available as VST or AU, as is the case with a lot of the little one man development teams that either can’t afford, can’t get approved, or simply have no desire to acquire Avid’s SDK. Using FXpansion’s VST to RTAS wrapper is necessary but seems to be working fine with all of the plug-ins I’ve tried so far. Many of them induce a mean latency that causes the delay compensation in HD to go nuts, so I generally have been using them directly on audio tracks before any TDM plug-ins or on the mix bus with the delay compensation manually disabled (it’s the mix bus, everything is going through it so it doesn’t need delay compensation). Unfortunately, dealing with Pro Tools HD’s piss poor delay compensation system is something I am all too familiar with as the UAD system I use creates some mean delay problems, especially when you start using them on sub-mixes and routing second order sends. Seriously, Avid needs to give us more than 4000 samples of delay compensation, which was cool like 6 years ago when it came out, but that is just not enough for the plug-ins on the market these days. I’m totally willing to forgo some extra DSP if it could be bumped up to over 8000 samples.
At any rate, you should definitely check out the stuff Stillwell is coding, a lot like the Massey gear it is super fresh and costs almost nothing.
This entry was written by , posted on May 10, 2010 at 1:01 pm, filed under News and tagged limiters, plug-ins, stillwelll. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
So I’m lurking around on one of the more notorious audio message boards (you know the one, rhymes with “deer butts”) and I am checking out some of the buzz over an unreleased plug-in currently in development (I wont mention the plug-ins name because I have no gripe with the developers and actually think it could be an interesting piece of software). Anyway, the usual listening tests, commentary and e-peen measuring ensued, the trolls came out with their classic “null tests,” everything played out as expected, business as usual.
The bulk of the responses to this new plug-in were surprisingly positive, like I said, I think it could be a cool addition to a mixers toolkit but I won’t know for sure until I actually try it. But what amazed me the most was the number of posts from people fretting over the pending release date and how they would have to hold off on all their mixing projects until it came out, and we’re talking weeks not days. Site unseen, holding back mixes for something that isn’t even available to demo because of two audio samples created by the company that is selling the product sound, “pretty good.” Are you serious? I can’t believe how people continue to not get it, believing that their mixes suck solely because of their lack of some “magic bullet,” that once obtained, will allow them to transcend space-time and instantly become a better mixer; a mixer with taste, vision and a command of esthetics. By the way these posts we’re laid out you’d think Pfizer was in clinical trials with a pill for creative inspiration, “Creativia® - the viagra for your other brain.”
I could understand the first few times an amazing new plug-in or technology was announced that people might adopt this mentality, but common, the industry has been getting over on audio engineers for years with this kind of marketing bravado. It is almost like the latest plug-in is advertised and hyped like it was the latest weight loss scheme, and people are so afraid of accepting reality. Writing, recording, producing, mixing and mastering great music takes a lot of trips to the old wood shed. Practice, patience, and time to develop sensible tastes and understand the big picture. If a musician told you, “I have the most incredible song in my head, but I’m waiting for my new guitar to ship before I work it out,” you would probably laugh at them. Better yet, if an unhealthy, overweight friend told you, “I’m waiting for this new diet book to be released next month before I start trying to lose weight,” you would probably just shake your head in disbelief. Waiting to work on a mix for a plug-in you haven’t even demoed, let alone learned and integrated into your workflow is just a shame. Placing these kinds of artificial limitations on yourself will only perpetuate the idea that there is a magic bullet out there and you will end up just waiting some more when the next plug-in in announced.
Reading these posts reminded me of a story my favorite mastering engineer told me the last time I was in a session with him. He told me that he gets a call at least once a week from some amateur fader jockey claiming that the only reason they need to hire him is because they don’t have the quarter million dollars worth of gear that he has. These callers place so little value in the human element a mastering engineer adds to the point of being verbally confrontational. This made me so sad, but I totally believed him. It is not hard to believe, especially here in the silicon valley, that there are purely logic-centered people who feel that music engineering, especially mixing and mastering, is a completely objective set of skills and can be understood like any other software program or hardware schematic. This group insists that simply by having access to the tool and the operating manual will allow them to get the results they desire in a very short amount of time. It’s the “hell, if I can program this software, using it to make better music should be a piece of cake” mentality of many of these aggressively logical, tech savvy folks that tends to perpetuate this type of behavior (hint: they are generally the same types that spend more time posting on forums than actually working on music).
The reason I wanted to share these insights is not to belittle or berate this kind of mindset, but hopefully shed some light on the mistakes that I made so many times earlier in my career. You see, I was that dude who would hang on every software release, and pine for the latest audio hardware, pre-amp, or mic. I figured, shit, after programming my own modem drivers for linux, how hard can this mixing stuff be? Maybe it is a necessary part of growing into your own skin as an engineer, but a part of me wants to think that if someone politely tapped me on the shoulder and said, “hey kid, you wanna know the big secret? It’s practice, time and patience, and even then you wont know everything.” There is no doubt that great gear, both hardware and software, is a necessary part of any professionals toolkit, but at the end of the day they are just tools. Tools that provide incremental improvements to your workflow as you acquire the knowledge and experience to implement them into your work. Becoming a better mixer, producer, songwriter, or musician is an incremental and iterative process rather than a sudden paradigm shift in understanding, courtesy of some magic bullet plug-in, tip or tool. The sooner you understand this the sooner you can get on with your life and start worrying about the real magic element in art, the human element.
So my humble suggestion is, don’t wait for that plug-in, microphone, pre-amp, guitar, or *insert piece of audio gear here* to work on your art. Do yourself the favor of giving your skills and ears the benefit of the doubt. Take control of you own destiny and own up to the results, good or bad. Remember, the path to excellence in any artistic endeavor is never ending and uniquely different for everyone, don’t let a corporate road map of product releases sully the journey .
This entry was written by , posted on April 20, 2010 at 3:26 pm, filed under Articles, featured, News and tagged mixing, plug-ins, rant. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
Just did a write up on the new Waves’ Vocal Rider plug-in for audioMIDI.com, check it out.
This entry was written by , posted on November 19, 2009 at 12:13 pm, filed under News and tagged mixing, plug-ins, waves. 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 November 4, 2009 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.
So I use quite a few “wrapped” RTAS plug-ins, meaning that the plug-in was designed for VST and hacked (via FXpansion’s VST-RTAS wrapper utility) to work as RTAS in Pro Tools. For example, all of the UAD stuff shows up as wrapped plug-ins in my inserts menu. This isn’t a big deal, as with most VST plug-ins the wrapper utility works flawlessly, but no matter which plug-in you end up using, it hides itself in the “Wrapped Plug-ins” sub-menu of your inserts menu. I have talked to Avid (Digidesign) about this and unfortunately, as long as they are “wrapped” VST plug-ins there is no way to get them into their correct categories (e.g. EQ, Dynamics, etc), aside from possibly hacking the plug-in .dpm file (I’ve looked into this. Without some sort of hint as to how the category identifier is determined/stored, I can’t see anything obvious when I pull up the files in a hex editor). While there is currently no easy way to get the wrapped plug-ins into their correct catagories, there is an easy workaround of sorts that will get you to your favorite plug-ins quickly.
Default EQ and Dynamics:
The first thing you want to do is set up your mixer’s default EQ and Dynamics plug-ins, these will show up at the highest level of the insert selection menu so put your “go to” EQ and compressor plug-ins in these two slots. To set the default EQ/Dynamics choose Setup > Preferences > Mixing Tab.
Plug-in Favorites:
The default EQ and Dynamics option only allows you to save one favorite within the EQ and Dynamics sub-menu, which blows if you want use a wrapped plug-in as your favorite EQ or compressor. In this case, you can save a plug-in as a “favorite” and it will show up at the top of the plug-ins list (within the plug-in type sub-menu on stereo tracks, e.g. TDM/RTAS or multi-channel/multi-mono).
To save a plug-in as a favorite: Hold Command (Mac) or Control (Windows) while selecting a plug-in from the inserts menu. The plug-in will not be inserted but will be stored as a favorite. Remember, you must hold Command (Mac) or Control (Windows) before clicking on the track’s insert selector.
To remove a plug-in favorite: Repeat the steps above, hold Command (Mac) or Control (Windows) and re-select the plug-in stored as a favorite from the insert selector menu.
You can have as many favorites as you’d like however keeping this list small will ultimately save you more time.
This entry was written by , posted on October 19, 2009 at 12:12 pm, filed under Articles, MixTips and tagged mixing, MixTips, plug-ins, pro tools. Leave a comment or view the discussion at the permalink and follow any comments with the RSS feed for this post.
This article is an excerpt from my column, “The Pro Tools Corner” at audioMIDI.com
NOTE: This was published before automatic delay compensation was standard in every Pro Tools system. Although the concepts and techniques in this article still work, you’ll want to read the section below regarding using automatic delay compensation in Pro Tools HD, it now works the same in all PT systems now.
Pro Tools Delay Compensation – Part 2
In the last installment of the Pro Tools Corner I discussed the challenges of plug-in latency in the mixer and offered up some techniques for manually combating this latency in Pro Tools. While the tips and tricks offered up last week will get you through most situations, there is an additional built-in solution for users on Pro Tools HD called automatic delay compensation that can be a real time saver. LE/M-Powered users can also gain similar functionality through a 3rd party add-on from MellowMuse called auto-time adjuster. In this week’s Pro Tools corner, I will discuss these two additional techniques for combating plug-in latency and offer up a few more tips for keeping your mixes in sync.
Pro Tools|HD Delay Compensation: (note: this technique now works in all PT versions 9+, not just HD)
Unlike native systems, which generally account for most plug-in latency within the system’s playback buffer, HD system’s using TDM plug-ins almost always suffer from small amounts of routing delay due to the inherent nature of hardware DSP processing. While the complexities of the TDM processing infrastructure are way beyond this article, one should note that plug-in latency is a serious concern when using TDM plug-ins in Pro Tools HD, so serious that Digidesign finally added a comprehensive delay compensation engine in version 6.4 of the Pro Tools HD software. While I personally try to forget about the days before automatic delay compensation in Pro Tools HD, I am consistently surprised at the number of HD users that fail to take advantage of such a critical tool in the mixing process.
Enabling Delay Compensation:
Because every TDM plug-in instance in Pro Tools causes at least a few samples of delay, automatic delay compensation is really a mixers best friend and a near hands-off approach for dealing with plug-in latency. To activate delay compensation, you must first enable it for your system in the Playback Engine (Setup > Playback Engine). The “Delay Compensation Engine” can be set to either Short (for up to 1024 samples of compensation) or Long (up to 4095 samples of compensation). If a session is currently open, Pro Tools will automatically save, close and reopen the session to enable the delay compensation engine. Once enabled in the Playback Engine, you can turn delay compensation on and off by selecting Operations > Delay Compensation.
Short or Long?
The delay compensation engine should be set according to the type of plug-ins you plan on using versus the amount of DSP resources you want to dedicate to delay compensation. Certain plug-ins cause more than 1024 samples of delay (many real-time “tuning” plug-ins, pitch shifters, etc) and therefore would not be sufficiently compensated for under the “short” setting. The down side of using longer delay compensation settings is that it cuts into the amount of available DSP your system has for other TDM plug-in processing. If you have a smaller HD system (e.g. HD|1), I recommend setting the delay compensation to “short” and manually compensating for longer delays using the techniques described in the previous article. I personally use a HD|3 and because I consistently use plug-ins that create a significant amount of delay, I leave my delay compensation engine set to “long” and find that I still have plenty of DSP left for most mixes.
Viewing delay compensation:
The Pro Tools HD mixer has a special view just for delay compensation, access this by choosing View > Mix Window > Delay Compensation. Here you will find the tracks total delay (as accumulated by either plug-ins or bus routing), a manual offset control, and the tracks total compensation. “dly” and “cmp” are calculated and adjusted automatically, so for the most part you can watch it do its magic and mix away, knowing you are taken care of. These values are displayed in samples by default, but can be switched to millisecond in the preferences > operation tab.
The longest delay in the session is denoted in orange and if a track’s total delay exceeds the engines maximum compensation amount (1024 for short and 4095 for long), it will glow red. When a track’s delay exceeds the total amount of delay compensation available you should always disable that track’s compensation by Control+Command-Clicking (Mac) or Start+Control-Clicking (PC) on the word “dly” in the delay compensation view. This will disable the delay detection on the track and at that point the track can be compensated for manually using the techniques discussed in part one of this article.
Tips for using delay compensation:
• When a track (audio or instrument) is record enabled, delay compensation for that track is automatically suspended to allow for low latency monitoring. To compensate, newly recorded tracks will be automatically shifted earlier by the amount of the total system delay after each record pass. You can force delay compensation on any record enabled track by Control+Command-Clicking (Mac) or Start+Control-Clicking (PC) on the “cmp” field, the compensation value will light up in blue to denote this.
• The master delay compensation indicator in the edit window can be used to quickly identify if delay compensation is enabled and working correctly (when green). When delay on any track has excited the total amount of compensation available, this indicator will light up red as a warning.
• You can set up delay compensation for hardware inserts in Setup > I/O set-up > Hardware Insert Delays. You will have to manually ping your hardware inserts to figure out each ones unique delay (play a transient rich sample or sine wave out into the hardware insert and record it back into another track, measure the offset to find the delay).
MellowMuse ATA: A Solution for Pro Tools LE/M-powered
Update: This is no longer necessary in Pro Tools 9 and later, read the above section.
The basic idea of any PDC system is that all tracks in the session incur the same amount of delay, thus eliminating any timing or phasing issues associated with some tracks lagging behind others. How most systems (including Pro Tools HD) accomplish this is to calculate the delay of the track with the most latency and work backwards from there. Let’s call this longest delay “X” samples. Tracks without any latency (a track without plug-ins for example) will get delayed by X samples, while tracks with a delay less than X will be delayed by X – Y, where “Y” is the amount of delay on the track in question. When all tracks, including auxiliary returns, exhibit the same amount of latency, we have a happy mixer. Like I said, because Pro Tools LE/M-Powered doesn’t do this for us automatically, we can use the ATA plug-in to calculate and compensate for us.
While the instructional videos on Mellowmuse.com are super helpful when setting this up, here is a quick breakdown of how the ATA plug-in is used in Pro Tools:
1. First insert ATA as the first plug-in on every track of your session, including the Master Fader and any Aux Tracks.
2. Inside the ATA plug-in window, set the plug-in’s “group” selector to the type of track it is located on (Audio for audio tracks, Aux 1 for Auxiliary Returns, and Master for the Master Fader)
3. Submix all audio tracks into a new stereo aux track, again placing the ATA plug-in as the first insert on this new aux. This is necessary to compensate for delay incurred on the effects returns.
4. Make sure nothing is muted, soloed or set to -infinity, and click the “P” or Ping button on the Master Fader to calculate and compensate for delay.
5. If you are going to do second or third order sends (Using sends on your aux tracks) you will need to use “Aux 2,3,4,5″ groups to compensate for additional latency (e.g. feeding a delay into a reverb, or using a send on a submix). The video at mellowmuse.com really helps to understand this.
6. Be sure to ping the master fader each time you insert a new plug-in, as that additional plug-in may change the delay on that specific track.
Note: you will not see the “Dly” indicator change to reflect all tracks having the same latency. The delay is compensated for internally inside the ATA plug-in.
The cool thing about ATA is that it actually polls the delay using an audio “ping,” which provides a super accurate measure of delay (some plug-ins report their delay to the host incorrectly). I recommend setting up session templates that have the ATA plug-ins, submix and effects return routing already set up. If you follow the instructions, it works flawlessly and is way better than having to manually compensate for delay, but I’m not gonna lie, this is sort of a cumbersome workaround if you use a lot of second and third order sends in your mixes (seriously, set up templates). But if you are going to point fingers, it’s really not Universal Audio or Mellowmuse’s fault. Avid needs to come up with a PDC solution for its LE/M-powered users and join the 21st century like every other DAW has.
This entry was written by , posted on October 13, 2009 at 1:03 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.
This article is an excerpt from my e-news column: “The Pro Tools Corner” at audioMIDI.com
NOTE: This was published before automatic delay compensation was standard in every Pro Tools system. Although the concepts and techniques in this article still work, you’ll want to read part two, specifically with regards to using automatic delay compensation in Pro Tools HD, it now works the same in all PT systems now.
Combating plug-in delay in Pro Tools – Part 1
Consistently a hot topic on many DAW related message boards and a recurring subject in my email inbox, plug-in delay can be a mixers worst nightmare. From subtle “phasing” to flat out timing and sync issues, the processing delay caused by certain plug-ins can throw an entire session out of whack, sometimes with as little as a single instance of said plug-in. While most of today’s DAW’s incorporate “automatic delay compensation,” effectively solving these issues behind the scenes, Pro Tools LE and M-Powered do not currently provide a transparent solution for users. And although Pro Tools HD provides a comprehensive automatic delay compensation system that works quite well, I am consistently surprised at how many HD users do not know about, or understand how to use it effectively. In this two-part article, I will attempt to break down the challenges of plug-in delay and provide a variety of solutions for both Pro Tools LE/M-powered, as well as Pro Tools HD users. In part 2 of this article, I’ll even share with you a handy new delay compensation plug-in that may be the perfect interim solution for Pro Tools LE/M-powered users.
Understanding the problem
First things first, it is important to understand the difference between latency issues during recording vs. the problem of plug-in latency (delay) in the mixer. When recording into a native DAW, you will most likely experience a small amount of delay related to the interface and the host’s processing buffer. This latency is un-avoidable and is only an issue while recording real time into the DAW. Most of the time this latency can be managed with a variety of techniques, by configuring the playback engine’s buffer size. Today however, I want to discuss specifically the delay caused by certain plug-ins in the mixer, which when inserted, can cause tracks to become out of alignment which each other, leading to phase problems and in extreme cases, sync issues in the mix.
To better visualize this dilemma, allow me to layout a couple scenarios in which plug-in delay would create a potential problem:
“Look-ahead” plug-ins:
Lets say you have a pretty involved multi-track session and you decide to place a real-time vocal tuning plug-in (auto-tune, waves, etc) on one of your vocal tracks. Because the tuning plug-in must analyze the incoming audio and re-pitch it in real time, there is some additional processing overhead that must go, beyond the normal processing buffer available to real time plug-ins. As a result, the vocals will play back later in time, possibly so much so that they sound out of time with the rest of the un-processed tracks. This is most often an issue with real-time tuning plug-ins, “look-ahead” style compressor and most brick wall limiters (R-Compressor, L2/L3, Maxim, etc), drum replacement plug-ins (TL Drum Rehab, Drumagog), noise reduction plug-ins, or any plug-ins that are going to exhibit significant processing latencies within their algorithms.
Absolute phase coherency:
With multi-mic processing (drums, etc) and certain “double-bus” or “parallel” mix tricks, absolute phase coherency is a must. Lets say you have a vocal track that you have duplicated onto another track, running one un-compressed and the duplicate completely squashed in a “parallel compression” style set-up. If the compressor plug-in you use on the squashed track induces even a single sample of delay, the two duplicate tracks will experience a phenomenon known as “comb-filtering,” as certain frequencies are canceled out or amplified as a result of destructive and constructive interference between the original and slightly delayed signal.
To better demonstrate, try this experiment for yourself: Take any track in pro tools and duplicate it (Track > Duplicate). Now place a plug-in known to cause delay (like Waves R-Compressor or L1) on the duplicate. Play back the session, muting and un-muting the duplicate track, the phase cancellation should be more than obvious.
Why it’s not THAT bad:
At this point many of you may be gripping your mouse with sweaty palms, wondering how you ever mixed anything down with such a monster delay problem happening behind your back. Well don’t fret, because in a native system like Pro Tools LE or M-powered, plug-in latency is in many cases, non-existent (Pro Tools HD is a totally different story and we’ll talk about that next time). The truth is that the Pro Tools LE/M-Powered mixer and most RTAS plug-ins exhibit zero delay, beyond the unavoidable delay induced by the H/W playback buffer that all tracks suffer from equally. Furthermore, even if the plug-in in question does induce delay, it is usually negligible (under 64 samples) and is only really a concern with phase coherent tracks (described in the second scenario above). Seriously, if your bass track is 4 samples later than your vocal track (less than a tenth of a millisecond at 44.1khz), it really doesn’t matter. Most of the time we are talking about microseconds here folks, so again only in phase coherent situations or when using super latent “look ahead” plug-ins do you really need to worry.
But how do I know for sure?
Fortunately, there is a really easy way to check for plug-in or mixer induced delay on Pro Tools tracks. From the mix window, simply Command-click (or Control-click on PC) directly on a tracks Vol/Peak/Delay display (located beneath its volume fader, right under the track type icon) until it reads “dly” (for example, you will need to command-click twice if “Vol” is displayed). This value is displayed in samples, you can convert samples to millisecond simply by taking your sample delay, dividing it by your sample rate and multiplying it by 1000 (so if you delay was 128 samples at a sample rate of 44.1Khz that would be 128/44100 x 1000 = 2.9ms Basically, if delay reads “0,” then you have nothing to worry about. If the delay shows a number like “64”, then you have a few options. While some specific plug-ins do misreport delay times, this is not super common, in fact some emulations introduce band specific delays that cannot be compensated for completely (again rare), ultimately you use ears as the final judge and make sure to comb the user’s manual if you believe that a plug-in is misreporting.
Hint: you can Option+Command-Click (or Alt+Control-Click) to display “dly” view on all tracks.
What to do if a track exhibits plug-in delay:
As I stated earlier, usually you don’t have to do anything if the delay is either a) not part of a phase coherent track grouping and b) less then 64 samples, as any more delay might cause timing discrepancies between rhythm tracks. Sometimes you can get away with delays of up to 256 samples on non-rhythmic tracks (vocals, etc), but this is up to you so trust your ears. Another case where no intervention is required is delay on the master fader, as long as all tracks are being sent out a common stereo output (e.g. your not using a summing mixer) you can get away with almost any amount of plug-in delay on the master fader since all tracks will be equally affected by this delay.
If for some reason you find yourself in a scenario where plug-in delay needs to be accounted for you have a variety of manual compensation options in Pro Tools LE or M-powered:
Shifting or Nudging tracks:
In scenarios when the plug-ins algorithm causes significant delay (tuning plug-ins, drum replacement, etc), so much so that the latent tracks play out of sync with the un-processed tracks, you best bet is to nudge or “shift” the latent track backwards (earlier in time) in the edit window by the amount of the delay. By shifting a track earlier in time in the edit window, the track plays back earlier thereby “anticipating” the delay it is going to gather up through processing.
For example, if your tuning plug-in showed 1024 samples of delay in the tracks “dly” view (more than enough to sound out of time), simply select all the regions on the track (triple-click with the selector tool) and choose Edit > Shift. Select “Shift Earlier,” in the sample field type in the amount as displayed by the “dly” indicator and hit ok. If the tracks regions are butt-up against the beginning of the session you may have to trim a small amount of the head to make room for the shift.
Make sure to record this shift in the comments field of the track, something like “Shifted earlier by X samples to compensate for ABC plug-in” works great. This way if you change out the plug-ins, you can get back to square one. I like to create a separate playlist before I shift tracks by significant amounts, because things can get a little crazy while keeping track of multiple plug-in induced delays.
Using the Time Adjuster plug-in
With shorter plug-in delays, as is usually the case in phase coherent scenarios, the Digirack Time Adjuster plug-in can be used. Time adjuster is a simple plug-in that you can insert on non-delayed tracks to match the delay of the latent track to maintain absolute phase coherency. Time adjuster is found under the “Delay” category and on stereo tracks it is easiest to use the “Multi-Mono” version. Time adjuster comes in 3 flavors: short, medium and long. Short supports up to 259 samples of delay, medium up to 2051 samples, and long up to 8195 sample. Remember the time adjuster will always cause at least 4 samples of delay, so if a track is show a delay of 3 samples or lower, you will have to do some nudging.
For example, lets say you have “Bass DI” and “Bass Cab” tracks from the same take, in other words they are phase coherent. After checking the dly indicator on each track you see that the DI track is being delayed by 64 samples and the Cab track by 128 samples. The goal here would be to use the time adjuster plug-in on the least latent track, in this case the DI track, to add more delay. By adding the time adjuster plug-in to the DI track and setting its value to 64 samples, I now show 128 samples of delay on both tracks, and have maintained absolute phase lock between the two.
The goal is to match the amount of delay so that each track in the group is delayed by the same amount. In this case, a total delay of 128 samples (roughly 3 ms) on the bass tracks would probably not warrant the adjustment of other tracks, but if it did affect the groove I might opt to shift the tracks in the edit window using the previous method. In other words, I would not place the time adjuster plug-in on every track to compensate for a tiny bit of delay on the bass track, this will just eat up DSP and I would have to modify every time adjuster plug-in in the session each time I messed with the bass plug-ins.
Sometimes for a quick fix I will just Option-Drag (Alt-Drag on PC) to copy the same plug-in to each track, bypassing the plug-in on tracks I don’t want to process. If each track has the same exact plug-ins, they should exhibit the same amount of delay (although at the cost of extra processing for a plug-in you technically aren’t using).
Tip: If you are using hardware inserts, you will need to compensate for the I/O latency in addition to any latency exhibited by the hardware device.
In conclusion:
Obviously you can see how all of this could get a bit tricky, juggling different delays on different tracks and modifying time adjuster as you add/remove/change plug-ins, which may make you think twice about using delay inducing plug-ins on phase coherent tracks or avoiding them entirely if possible. Ideally the software would calculate the delay on each track and delay all other tracks automatically so that each track exhibited the same amount of delay at the end of the chain (Other DAWs and Pro Tools HD will do this for you).
Now before you go out and hate on Pro Tools LE/M-powered for not having automatic delay compensation remember, your frustration may vary based on the types of plug-ins you use. Many users will never experience noticeable problems, in fact if you stick to the Digirack plug-ins you are good to go, as they induce no extra delay on the mixer. Fans of look-ahead style processors, which includes any kind of decent limiter plug-in, will suffer a bit more as these almost always cause delay no matter what flavor you use. Like I said, as long as no track is showing over 64 samples of delay (sometimes you can get away with 128) and your phase coherent tracks are taken care of, you should be fine.
Coming up in part 2:
In part two, I will show you how to use MellowMuse’s ATA plug-in, a clever solution for the delay compensation problems in Pro Tools LE and M-Powered. I will also run through Pro Tools HD delay compensation and explain why it is a must use feature for HD mixers and producers.
This entry was written by , posted on October 10, 2009 at 10:13 am, 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.
