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I have been very busy lately with many different (and great) projects going on. I wanted to update you that I just finished mixing an album by Go Go Ghost, called Summer Stunner.
You can listen to it on- gogoghost.bandcamp.com

I have many ideas for blog updates in the near future! I’m working on a series of articles and videos on microphone techniques. If you want to be updated when the new posts are ready, you can sign up to the mailing list by clicking here (or on the top right of the blog).

* also, I updated the interview videos with Al Schmitt to youtube videos, so if for some reason you couldn’t see them before, you can now.

A very big difference between Pro Tools HD and LE (this also includes M-Powered) is the issue of delay compensation. In Pro Tools HD you can activate “automatic delay compensation” and the TDM engine will compensate for latency introduced to each individual track. Pro Tools LE (including M-Powered) does not have this feature.

Crash Course:

TDM stands for Time Division Multiplexing. Pro Tools HD is a TDM system. This refers to the method by which the signal processing is performed on PCI cards rather than by the host computer which is running the program. LE and M-Powered systems are called “Native” systems, where all of the digital signal processing has to be handled by the CPU of the host computer that is running Pro Tools.

Audio Plug-In – a plug-in is a program that runs within a host-program. It is inserted in a digital audio path and performs a mathematical process (known as an algorithm) to the bits that represent the audio. Examples of plug-ins are EQs, compressors, delays, reverbs…

Plug-In Delay or Latency - Some plug-ins have algorithms that introduce a short amount of delay from the time the audio “enters” the plug-in on one end and the processed audio “exits” the plug-in on the other end. This short delay is commonly referred to as latency. An example for such a plug-in is a look-ahead plug-in such as certain peak limiters or a real-time vocal tuner that has to delay the output by a certain amount in order to “look ahead” before it processes the audio.

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Latency can also occur by using sends to bus audio from one track to another, usually for the purpose of effects such as reverb or parallel compression. The short delays introduced in this way can have a very small effect on the outcome (such as reverb sends). Other times, the latency can create artifacts that are very audible, to the point where you cannot even use these effects in native systems (for example when you try to use sends to do parallel compression, which we will get to in a future post). If you have two very similar tracks, the introduction of delay to one of those tracks can create phase cancellations and cause a comb-filtering effect like a flanger.

In cases where these short delays may not be very audible, they still cause your mix to lose clarity and punch. If you were mixing in the box you may not notice this latency too much, but if you heard that same mix without the latency – the mix would instantly gain clarity, definition and punch. I bet you would choose to use the delay-compensated mix .

One way to get around this issue is to manually nudge each track to compensate for the delay. In the Pro Tools Mix window, at the bottom of the fader there is a volume readout. If you command-click (or for you Windows users, control-click) on that “vol” readout once, it will change to a “peak” readout (telling you what the highest meter peak was on that track). If you click the same spot a second time, you will get a delay readout labeled “dly” which is labeled in samples. In this case, the sax track, which has several plug-ins inserted on it, has a delay of 64 samples.

We would now have two options: We would either have to go to the edit window, select the region (or all of the regions in the case of edits or multiple takes) and nudge them to the left (earlier) by 64 samples. The second choice would be to insert a “time adjuster” plug-in on all of the other tracks and delay all of them by 64 samples. This is only if the sax is the only track with any delay on it. If more tracks have different delay values (which could very well be the case) it can get complicated to manage. Add to that the scenario where you may want to continue inserting or removing plug-ins on the sax track or any other track, and then you have to go back and nudge the regions again, or change the time adjuster values (for all of the other tracks). As you can see, it can be a real pain.

I have recently learned about a great plug-in that is specifically designed to solve this problem of track delays. It was designed in a smart way that can handle delays caused by just about any reason, and it is fairly simple to get a handle on its operation.

Mellowmuse ATA

The plug-in is made by a company called Mellowmuse and is called ATA, for Auto Time Adjuster. I will provide an explanation here but if you want to watch some video tutorials and read about it directly on their site, you can go to http://www.mellowmuse.com/ATA.html

The basic idea behind this plugin is that you insert it in the first insert slot on each track, before any other plug-in. You have to designate the type of track (audio, aux or master) and you need to have a master fader with the “master” designated ATA plug-in inserted in the session. Once this is set up, you open the “master” plug in and hit the “P” button. This stands for “ping”. This will cause the tracks to “ping,” with each track firing away individually. The master picks up these signals and measures the time it took to reach the master bus. That way it calculates how much latency each track has, and automatically compensates for it. Pretty elaborate, isn’t it? If you added, removed or changed any plug-in, simply hit the “P” again and it will automatically recalculate. Easy!

There are a few things to understand before using this system. First, all of the tracks in the session have to be un-muted before activating the ping. The plug-in does not “understand” when a ping signal never makes it to the master, and you will start hearing some tracks shifting around.

The second thing that is important to understand is the setup hierarchy with aux tracks. The “group” I was talking about earlier – audio, aux1, aux2, aux3, master – refers to the type of track on which you insert the plug-in. Something a bit confusing here is that when you set it on an “aux” track you must understand that ATA refers to “levels” of auxiliaries. Meaning, let’s say you have a plate reverb on Aux 1, a room reverb on Aux 2 and a delay on Aux 3. in Pro Tools, they are called Aux 1, 2, 3… but to the ATA plug-in, these are all seen as the first level of Aux tracks. Let’s say that the session is more complex and you decided to send all of the drums to Aux 11 (let’s say it’s mono, for explanation purposes) and all your guitars to Aux 12. Then you decided to send the signal from Aux 11 (the drum bus) to the room reverb and the signal from aux 12 (the guitars) to the delay. In this case, the drum bus (Aux 11) and the guitar bus (Aux 12) are the first level of Aux tracks and you would set ATA to the group “Aux 1″ on those tracks. The reverb and delay in this case would be considered the second level of Aux tracks, and ATA for these would need to be set to the “Aux 2″ group. Think about the stages of latency here and it will help you understand what the designers of this plug-in meant. When you open a send from an audio track to an aux track there is some delay introduced to that signal. That would be the first level and would refer to the “Aux 1″ latency group. If then you send a signal from that aux track to a second aux track, you are once again delaying the signal a little bit, and that is the second level which would refer to the “Aux 2″ latency group.

There is a small amount of setup to do here, and at first you might need to get used to the concept but once you’re set up all you have to do is hit the “P” (=ping) button and you’re set. Another good thing is that since the plug-in uses ping and not just a latency readout from the plug-ins, it is guaranteed to reflect the true delay value of the track. Also, the plug-in should be able to accurately compensate for any hardware inserts as well. I have not tried this yet, but I will soon try it using outboard EQ when mixing. I will update you and tell you how it worked!

Best of all, they let you download a demo to make sure it works on your system, and for you to learn how to use it. It works with many different plug-in formats, so it is not restricted to Pro Tools – you can use it with almost any audio software. The best part is that it is not an expensive plug-in – the full version costs $49. I highly recommend it!    http://www.mellowmuse.com/ATA.html

Here is a video with legendary audio engineer Al Schmitt. He is interviewed by Claris Dodge from studioexpresso.com. (This video was produced for recordproduction.com)

Al Schmitt is an amazing engineer and a great person. I was fortunate to learn from him by attending his sessions at Capitol Studios in LA. I got to witness some incredible recording and mixing sessions with some of the most talented artists and performers around (and of course the talent of Al Schmitt and the amazing studios at Capitol). Watching Al work was a great experience, which I take with me to each recording session. He was always willing to answer my questions and share his techniques.

In this video he gives some valuable advice. You should pay attention, “read between the lines” and try to pick up everything that is said and unsaid about the recording process and the industry. I’m happy to bring you behind the scenes and get to see Al Schmitt speak about his experiences. Enjoy!

Do you know Sonic Scoop? www.sonicscoop.com

This site is run by my friend David Weiss, and it covers the audio industry in the New York area.

It’s a great site and I recommend you check it out, especially if you live in New York, and also if you don’t…

Enjoy!

~Mor

P.S. – Here’s a good article to start with. I really liked it – The Strokes Still Rule the Decade: “Is This It” Explained by Gordon Raphael

If you haven’t yet read the first part – LOUDER IS BETTER! (Part 1) Please scroll down and read it first, to better understand this article.

Now that you’ve had a look at the Fletcher-Munson equal loudness contours, you have an understanding of the way we hear the different frequencies. Let’s review the important concepts to understand here.

First- if we want to hear different frequencies across the audible spectrum at the same level, we actually have to play some frequencies louder than others. Our ears are most sensitive to the “presence” frequencies: 2-5 KHz, where out speech is most intelligible.

Second- and perhaps most important, the louder we play these sounds; the less sensitive we get to the differences in frequencies. This means that when we play music very loud, we will hear it with enhanced low end and high end content. When we lower the volumes, it will sound as if the low and high frequencies are being gradually cut off. Do you know those buttons on stereo receivers called “Loudness” ? What this button is actually meant to do is compensate for the loss of frequencies when listening at low levels, for instance at night when you can’t turn the level up to “normal” levels. Many of you, I’m sure, like to use this button even at moderate to loud levels, because it’s fun to listen to music rich in high and low frequency content! However, the true meaning of the loudness button is to maintain the correct frequency response of the system at lower levels, i.e. to compensate for the Fletcher-Munson loudness curve.

Let’s do an experiment. Below is an audio clip of a frequency sweep. All of the frequencies are played at the same level. (This is a tough experiment to conduct in this setting, as there are so many variables because each of you have a different system with different characteristics and abilities. Regardless, let’s try this. I recommend that you listen with headphones. This will hopefully reduce the variables a little bit. Also, there is a greater chance that your headphones will be capable of reproducing a larger range of frequencies.) Play the frequency sweep, and remember to keep the level low. As you have learned, the level at which you listen has a large effect on your perception. Try to see if you can hear how the level you hear changes as the frequency sweeps. Don’t worry if you find it is hard for you to tell the difference. In my opinion, hearing level differences is one of the hardest things to learn. It took me the longest to develop this sensitivity. It’s good to start thinking about this and sharpen your ear’s sensitivity to level differences.

frequency Sweep 20-20000-20s

Now for some conclusions:

- Having talked about the Fletcher-Munson Equal Loudness Contour, we can understand that if we mix at excessively loud levels, our mix won’t sound the same when we bring the levels back down. Our mix will become “small” as the high and low frequencies begin to roll off.

- If we mix at moderate to low levels, turning the level up will make things sound better, as we typically like to hear music rich in high and low frequencies.

- We want our mix to translate as best as possible to as many systems out there, and there are somewhat “established” standard for mixing levels. Typically, you should not exceed 85 dB SPL (more on what dB SPL means on a future post). This is mostly a standard when mixing for film. It is much easier to establish standards when you can somewhat control the environment the audio will be played back in (i.e. the movie theater). Generally, the recommendation is to stay within the 65-85 dB SPL range.

This is also good practice to protect your hearing. Every once in a while, I turn my mix up for just a little while, to hear what it sounds like at higher levels. I never leave the music loud for too long, however, as my ears will reach fatigue very fast that way.

As always, I would love to hear what you think in the comments section. If you want to be alerted when new posts are made available, sign up at the top right of the blog (www.EarsandGears.com)

Rock on!