Elastic Audio In Pro Tools – Part 1

This entry was written by Brian, posted on October 14, 2009 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.

Master Faders Demystified – Part 1

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:

1. Turn your monitors all the way down and please take off the headphones!
2. Create a new mono aux track and insert the “signal generator” plug-in (found under “other”)
3. Duplicate the new signal generator track 50 times (Track > Duplicate)
4. Enable the “ALL” group and turn one of the aux track’s level controls up to 12dB (all should rise to 12dB)
5. Carefully turn up your monitors just a bit, listen to the extremely distorted sine signal.
6. Create a new stereo master fader set to your main outputs (Track > New)Trim down the master fader until it stops clipping, you may have to clear clip indicators Opt + C (Mac) Alt + C (PC).
   

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.

Read Part 2

This entry was written by Brian, 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.

Master Faders Demystified – Part 2

This is an excerpt from my column, “The Pro Tools Corner” at audioMIDI.com. Read part 1 here

Update: Pro Tools 10 introduced HDX which uses a 64bit floating point mix bus and 32bit insert chains, effectively making it sonically identical to a native system. Older HD cards will continue to use the 48 bit fixed mixer and 24-bit insert points until they are not supported in Pro Tools 11. I also added a new section on native vs. TDM insert points.

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 2

In the previous installment of the Pro Tools Corner, I discussed the importance of integrating and correctly using master faders in your mixes. While master faders are quite easy to use and understand in the Pro Tools Native, HD Native, LE and M-powered mixers, using them effectively in Pro Tools HD TDM is an entirely different story. In part 2 of master faders demystified, I will cover some of the unique signal flow considerations HD users must account for while using master faders and also give you some overall pointers for using a master fader’s inserts for mix bus processing. If you haven’t already read the first part of this article, I suggest you check it out before proceeding.

Note: Remember I am talking about the TDM mixer here, if you are using HDX or Pro Tools HD Native, this does not apply in the same way, make sure you read part 1 of this article to effectively use master faders in your workflow.

Master faders in the HD TDM 48-bit mixer

First off, if you are a Pro Tools Native, HD Native, HDX, or older Pro Tools LE or Mpowered user, the following only applies if you plan on creating mixes that translate to a Pro Tools HD TDM environment. Otherwise, I assume you are reading this section from an HD TDM user’s perspective. If you plan on only using Native, Native HD, HDX or LE/Mpowered from start to finish, you should definitely check out the first part of this article for some useful insights on setting up and using master faders in your mix.

So what’s the big deal with the HD TDM mixer anyways? In the Pro Tools HD TDM mixer, think of tracks as having a 24-bit input and a 48-bit output. That is to say that the HD mixer adds or “sums” at a 48-bit fixed precision and has tons of headroom to combine many hot (near full code) signals without clipping internally. Now just like every other DAW on the planet, the end result of the mixer’s summation has to eventually dump out to 24-bit converters, creating a unique challenge for in-the-box mixers. The master fader’s job, just like in the Pro Tools Native mixer, is to act as a final output trim, or “bridge,” between the internal higher resolution 48-bit mix bus and the lower resolution 24-bit converters. When signals inside the mixer combine to exceed the maximum level allowed by the 24-bit outputs (which they often do), the master fader can be used to trim down the end result, thus avoiding any clipping at the converter. Because internally these signals combine at a very high precision, no detail is lost nor samples clipped when signals are added together inside the mixer. The master fader acts as a sort of bit selector, allowing the mixer to exit the system with the most significant or “best” 24-bits to the converter, not clipping, nor leaving too much excess headroom.

Now this is all fine and good until you start submixing into busses or using sends. Because of the way the TDM mixer and DSP chips are structured, every time you use a send or output set to a bus, a new unique 48-bit mixer is created for that bus path. This mixer lives as a separate entity to the main mix bus summing out to your converter’s outputs. Because the mixer is a separate entity, the signal is truncated back to 24-bits when the bus is “returned” to the main mixer via an axillary track (remember, tracks have 24-bit inputs and 48-bit outputs, this included aux tracks). The problem with this structure is that you can clip a bus’ mixer without clipping the main outputs, or in other words, you could be clipping your mix without lighting up the clip indicator on your main output’s master fader.

You can see in the screen shot that the submix is clearly clipping while the master fader is not. Even reducing the level of the aux submaster does not illimate the clipping because it is occuring on “Bus 1-2″ as it re-enters the mix on “Aux 1″ before the aux tracks fader can attenuate it.

You can see in the screen shot that the submix clipping has been eliminated by a new master fader set to the same bus pathway (bus 1-2).

Now as I said in last week’s article, this doesn’t happen the same way in the Native mixer because the mixer is 64-bit float through the mix bus and 32-bit float from insert to insert, meaning you can effectively use just a single master fader set to your main outputs (usually A 1-2). But if you want you mix to translate effectively to an older HD TDM system, you may want to follow some of these practices as a general habit.

HD TDM vs Native Insert pathways:

In Pro Tools HD TDM, insert pathways are 24-bits, just like bus inputs, so you can clip inserts without clipping the mix bus. How? For example, if you take a signal that is peaking at -6dBFS and you apply an EQ to it at one of the insert points and proceed to boost 12dB on any frequency, you will effectively clip the 0dBFS output by 6dBs. While most plug-ins have internal guts that process in floating point to prevent internal clipping of the algorithm, in Pro Tools HD TDM plug-ins, the signal will always be truncated back to 24-bits when it passes from one insert to the next, regardless of the way the internal guts handle overs. So if you were to preform that boost in your EQ, thus eating up all your headroom, and then brought that into a compressor and brought the signal back down to say -10dBFS, your signal would have already had the top 6dBs clipped off of it before you could bring it back down on the next insert. At this point, your master fader wouldn’t show clipping because signal would have already been clipped at the output of the EQ plug-in and subsequently brought down by the next insert. To avoid this, you MUST use an insert’s input and output trims when mixing in the TDM enviroment, even if the plug-in internally supports dual or triple precision processing.

The only caveat to this is if you are going from RTAS to RTAS insert (or AAX Native to AAX Native) in the TDM mixer. In this special case, the signal is indeed passed through to the next plug-in in the chain at 32 bit float. Try this experiment, take a sine wave at -5dBFS and run it into a TDM version of EQ-3, now do a boost of at least 10dB so you can hear the clipping distortion. Next place another EQ-3 TDM plug-in directly after in the next insert slot and pull the output trim down by at least 10dB. If both plug-ins are TDM, you will still hear the distortion created in by first insert clipping the output, even if you bring the signal back down to under 0dBFS in the second insert. Now take those two plug-ins and convert them to RTAS (or AAX native) using the icon at the top right of the plug-in (the one that says TDM). You should hear the clipping disappear, as the signal is now being passed at 32-bit float between the two plug-ins, thus allowing you to exceed the 0dBFS mark on one and recover that to below 0dBFS at the next insert point without any loss of signal.

Regardless of what happens with TDM and non TDM plug-ins in the 48-bit TDM mixer, in the Native mixer and the HDX mixer, RTAS and AAX plug-ins always pass their signals at 32-bit float, which would allow you to “carry over” any data exceeding 24 bits via the exponent into the next insert, leaving it up to you to make sure you reconcile your final signal at the master fader before hitting the D/A converter. However, I find that certain plug-ins (try Maxim for example) will truncate at 0dBFS regardless, based on the way they were coded, while other plug-ins, like vintage modeled compressors and EQs, tend to have a sweet spots based on their analog counterpart that may not sound as good as you’d like if you are pushing a huge amount of gain through it. Remember, most analog gear is set up to work best with a signal level around 0 VU, and most vintage modeled plug-ins are calibrated at between -14 and -20dBFS = 0VU, so keep that in mind when deciding how hot you want to run signals through insert chains. Bottom line, I find that gain staging still matters, even in a floating point environment.

It is my hunch that when I hear these guys saying, “I bought the new HD native card, or HDX card and my old mixes just sound so much better instantly,” that what they are hearing is either the sound of the new HD series converters (which sound a lot better than the blue faced ones) or they are all the sudden hearing there mix without insert and bus clipping for the first time. In reality, if you respect your gain stages and follow the operating rules of the system you are mixing in (be it TDM or Native), the mixes will be BIT for BIT the same. However, even tiny amounts of digital clipping happening over many insert points and busses can add up to a certain haziness that eats up fidelity, and muddles yours stereo image.

This “micro clipping” all over my mixer is why I am always super vigilante when mixing in the HD TDM environment. Because of this burden of always having to pay attention to every gain stage like a neurotic hawk, I have since moved over to doing all my mixing on using the Native mix bus, and only boot up the old TDM system when I need to do low latency tracking. With today’s super powerful computers and the fact that most of the cool new plug-ins are native only, I find that forgoing the TDM cards come mix down and mixing natively actually allows me to do bigger mixes with less trouble. Once more plug-in developers start supporting the AAX format, I might consider upgrading to the HDX system to get the best of both worlds, floating point mixer and DSP power to spare, but unfortunately I can’t just drop everything I’m working on and upgrade to an unsupported rig just yet.

Master fader inserts

In Pro Tools, master faders feature 10 post-fader inserts. Unlike regular inserts, which are pre-fader and do not take into account the channels volume control when processing a signal, a master fader’s inserts happen post trim and are, in essence, the final stop before the mix leaves the system and hit the converters. This makes master fader inserts the perfect place for master bus processing (or processing all the audio of a mix simultaneously). So on a master fader it goes Fader > Inserts > Meter.

The master fader is typically the place to insert a mastering, or “brick-wall” style limiter along with any dithering algorithm used for bit-depth truncation. Here are some tips for using a master fader’s inserts:

• Read the white papers: “Mixing in the Box” (http://akmedia.digidesign.com/support/…/Mixing_in_the_Box_26689.pdf) by Stan Cotey and “The Pro Tools 48-bit Mixer” (http://akmedia.digidesign.com/support/docs/48_Bit_Mixer_26688.pdf) by Gannon Kashiwa
  
• When submixing through a master bus internally, with the goal of printing your mix to a new audio track, you must set your master fader’s output to address that bus if you want its plug-ins to make it onto the new audio track. This does two things: first, it protects the new recording from clipping the new 24 bit audio file that is being created, which unless you are using Pro Tools 10′s 32-bit audio file format, will clip just like your converter. Second, it allows you to bring along any master bus plug-ins you are using into the printed submix. Otherwise you would need to first print your mix internally and then use “bounce to disk” to gain any additional processing reflected on your converter output’s master fader.
• When using dither on the master fader, it should be the last insert in your processing chain, as adding or subtracting any gain post dither will affect its placement at the least significant bit and negate its goal of reducing quantization artifacts. You should not use dither if you are going to print your mix internally and export the file from the regions list at 16-bit, as dither is automatically added to truncated files at export (currently, this cannot be disabled)
• When using “brick-wall” limiters/processors on your master fader, first make them inactive to ensure that you are not clipping the output or bus. In Pro Tools HD TDM, an insert has a 24-bit input and output (regardless of whether or not the plug-in itself does dual precision or 48-bit internal calculations) so they CAN and WILL clip. Normally plug-ins will identify input/output clipping by lighting up in red, however I have found that certain brick-wall limiters are not set up to show I/O clipping correctly. Therefore, you may unintentionally feed a signal from a clipped mix bus into a mastering limiter, set its ceiling to something lower than 0dBFS and never know that you mix is actually clipping the 24-bit inputs of your limiter.
• You may want to use the included TL Mastermeter plug-in to track when and over how many samples your mix clips at the master fader. This can help you can track down specific transients/sections that are causing trouble and attenuate them manually.
• While I might get rocks thrown at me for saying this, generally you don’t have to be hyper critical about a clip here or there. Being aware of the system and where clipping can occur is a step in the right direction. When you do decide to pay attention, you may be surprised at how much your mixes clear up.

This entry was written by Brian, posted on at 1:07 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.

Plug-in Delay Compensation in Pro Tools – Part 2

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 Brian, posted on 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.

Plug-in Delay Compensation in Pro Tools – Part 1

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 Brian, 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.