Leveling, meters, and many other topics along the way….

April 9, 2009

http://tarekith.com/assets/Leveling.html

Leveling, meters, and many other topics along the way….
v1.02

By Tarekith

One of the things that always surprised me in the past, was the number of DJs and musicians who didn’t understand what level meters were for, or how to use them. Walking into a club or bar and seeing the whole mixer lit up red used to make me cringe every time. Usually you didn’t even have to walk up to the mixer to see this, you could tell by how it sounds once you know what you’re listening for. When a friend said I should write a guide to help others understand this better, I realized it was something I wanted to do.

I’m going to be covering a few different areas in this guide, more or less walking you through the different aspects of DJing or producing in Live that might require your attention when metering is involved. Ironically, with a program like Live available, the tools and techniques that were previously only in the domain of one group or the other, now apply to both. I admit freely that I consider myself more of a producer than a DJ, so if some of this seems really anal to you, sorry, it’s just how I was raised :) I also admit that there’s certainly more than just one way to do these steps, so I’m sure many of you will disagree for one reason or another with some of my suggestions. So let’s just say up front that the opinions expressed here after are only my own, this is just the way “I” work.

So, on to the good stuff!

So, what IS metering, setting levels, and avoiding clipping really about? To put it simply, it’s about avoiding distortion. When an audio signal clips, you’ve exceeded the limits of something in your signal chain (all the devices that the audio flows through), or your storage medium. You are negatively affecting the audio, and metering was invented to let you know well before this happens.

Ironically, when it comes to using Live for DJing, the very first thing we need to look at when it comes to meters, is a standard DJ mixer. I know a lot of people out there were using vinyl to DJ with before they switched to a laptop, and most of you set about immediately recording your favorite records to use in your new toy :) From the start you need to be paying attention to metering, and when recording your favorite records, even more so!

So first, let me quickly suggest some good practices to use when recording vinyl. Because you can’t connect a turntable right to most computers or soundcards, we’ll assume you’re using a DJ mixer to record through. Our goal therefore, is to try and make the mixer as transparent as possible. So we’re recording as much of the MUSIC as possible, as closely as possible to the way it was intended to sound. I suggest:

– Clean your records first, and always use fresh needles.
– Balance your tone arms “properly”. You’re not going to be scratching, so keep it at the manufacturer’s recommendation.
– Don’t use a slipmat, if the record slips even the tiniest bit here and there, you will hate yourself when you’re warping it later.
– Set the mixer’s EQ to flat (centered), or better yet, turn it off if your mixer has this feature (i.e. Allen & Heath).
– Don’t route through the crossfader if you can, if not, make sure it’s centered and ideally new.
– Set your mixer’s channel gains to their center position. Don’t worry if it’s quieter than you’re used to. Everything will be when you’re comparing laptops to vinyl gear (and I’ll explain why).
– Make sure all FX sends and Aux’s are off or down, channel faders up all the way.

The record should now be able to play, without peaking the red LED’s on the DJ mixer’s master output. If not, lower the channel fader a little bit until it is not showing ANY red on the master meter. Don’t try and get the signal as close to red on the meters as possible, but it should be a medium, to medium strong signal.

Ok, now we’ve got the vinyl side ready to go, let look at our recording side. From the DJ mixer, we’re going to connect to the soundcard. If your soundcard has a software mixer that came with it, it’s a good idea to check your signal here next:

This way you can check that you’re not clipping the A/D convertor, the first step in our digital chain.

Huh?

Let’s stop for a second and make one point that’s very important to understand when it comes to clipping and leveling. In general, analog clipping will sound better than digital clipping. When you distort an analog signal, the degradation happens slowly, and sometimes with pleasant results, when used in moderation at least. This is not the case with digital signals at all. Distortion in a digital signal is all or nothing, on or off. It’s a finite limit that once you step over, the results are usually very bad sounding. So while you can get away with a little bit of analog clipping and it might go unnoticed, digital clipping is something we’re trying to avoid at all costs. Back to our signal path…

So, the nice clean analog signal enters your soundcard, where the A/D (analog to digital) convertors change it into a bunch of numbers, digital musical data. For our purposes, we’re going to think of these numbers as having two ‘dimensions’, frequency and volume. Frequency is governed by a digital signal’s “sample-rate”, which is simply how many times a second the analog signal was analyzed and recorded by the A/D convertor. Personally I just use 44,100kHz, which is the industry standard for CDs. By far the most common format you will run into, so best to stick to it.

Volume however is what we determine with a signal’s “bit depth”. Here we have a choice to make, before we even record a single vinyl crackle. Do we want to use 16bit (CD standard) or 24bit? To discuss both options’ advantages, let’s quickly get some simple math out of the way. This is something that we’re going to need to know later too, so don’t skip this part!

For all intents and purposes, 1 single digital bit, is equal to 6 decibels (1bit = 6dB). Therefore, a 16bit audio file has 96dB, and a 24bit file has 144dB of “space” they can use to record a signal’s strength, how loud it is in our case. This is called the dynamic range, the difference between the loudest and quietest audio signal we can record or play back. This does not mean that 24bit signals are louder, both files types are capable of the same volume. You need to think of the bits like this:

Corny, I know, but it beats freehand drawing blocks in Photoshop :) Anyway, as you can see, both files types are capable of the same volume, they’re level up on top. But the 24bit file can record quieter sounds, and this also has benefits in the math involved with ALL digital processing actions. Downside is that it takes more hard drive space to save these files, and more CPU power to process them.

So we know it’s going to be ideal to record using 24bits, but we’d like the benefits of the more widely used 16bit files. What I do is record my records initially using 24 bits, trim the ends, normalize to -0.2dB, then dither down to 16bit. If that made no sense to you, don’t worry about, it’s far too many things to explain here. Let’s instead look at a simpler way that has only a very, very slight difference in the audible end result.

Let’s use Live to record our record, and let’s use 16bit. You can change these settings in Live’s preferences, and it must be done before you start a new project. Once you set this though, it will remain for all new projects you open.

The next thing we want to do, is expand Live’s mixer section to show us version 6’s brand new, and MUCH improved meters. Drag the thin horizontal bar directly above the fader up as far as it will go, and drag the right side of the track header to the right some. This will show you the calibration marks and full-length of the meter, which are what we are really interested in. These numbers tell us how loud the digital signal is in a very accurate way, and are scaled using decibel (dB) numbering.

0 dB is the default (i.e. the track fader is not affecting the audio signal at all), and you can go up or down from there. Larger negative numbers being progressively quieter. So a 16bit file is represented by 0 to -96dB (16×6), with 0 being the loudest digital signal we can play back, and -96dB being the quietest. Live’s meters don’t even go below 60dB as you can see, but this is not an issue, as anything lower than that is really too quiet to be useful anyway. I know you’re wondering, well why does Live’s meter go to +6dB, if 0 is the highest we can go. Due to complicated maths (floating-point processing if you must know), Live’s track meters can actually give you a bit of “oops” room. However, I think it’s always a good idea to just ignore this safety net, and get into the practice of ALWAYS trying to avoid getting near 0dB with your audio signals. It’s easy to remember to do, if you do it all the time.

So start your turntable, and make sure you have armed an audio track in Live, and that you can see some sort of signal from your turntable:

If you’ve done everything right, your signal is probably pretty good already at this point. Don’t be alarmed if your signal is only showing in the middle portion of the meter, this is not a bad thing! Look at the pic above, notice how the meter shows only -12dB around the halfway mark?

Remember how 1 bit equals 6dB? This holds true here as well, which means that even at -12dB, we’re only losing 2 bits of possible resolution with which to represent our recorded digital signal. So our signal is now represented by 14bits, instead of 16bits. The key here, is to realize that we don’t even need 14bits to accurately record most records, especially if they’re dance tracks. Modern day music has had much of it’s dynamic range squeezed out of it via tools like compression and limiting. I’ll leave it up to you to decide what you think of this trend (I hate it, in case you’re wondering), but regardless of your view on the matter, the end result is that most of our music has a very small dynamic range. There’s much less of a difference between the quietest sounds in a song, and the loudest sounds. For dance records I’d guess that a good average is 6-12dB’s of dynamic range, for other genres maybe as much as 12-18dB if you’re lucky.

What this means is that we can likely record our songs using only 3bits these days, since 3bits = 18dB. Obviously that leaves no room for error, and there’s other issues that could arise using so few bits. But the main point here is that even with the file we recorded being only represented by 14bits, that’s still 11bits more than we’d ideally NEED to record the song completely accurately. So don’t worry if your meters are near 0 when using your computer programs, there’s nothing wrong with leaving yourself some extra room to stay away from that 0dB clipping point (this is called ‘headroom ‘ BTW). This is also why a lot of producers work at 24bit, it gives them 8 more bits (48dB) of room to play with.

Ok, so we’ve recorded our song, trying to keep the peaks of the recording somewhere around -12 to -18dB’s. Experienced producers may aim a bit higher, say -6dB, but you don’t NEED to do this to get great results. It’s easy to make a mistake here if you’re not careful, so stick with the side of safety and aim lower if you’re not sure. Messing up here will permanently distort your recorded signal!

The next step is to get the files ready to use and DJ with, and once again, meters play an important part. One thing I do, is normalize the files once they are recorded. It does technically degrade the audio a teeny, tiny, itsy little bit, but it’s worth it for me to have a nice large, clear waveform to look at while DJing. Helps me a lot with how I play, so the trade off is well worth it for me. Most people know I’m a freak about audio quality, so that tells you something about how small this degradation is hopefully. I personally use Wavelab and normalize everything to -0.2dB at this stage, but it can be done in Live too via it’s Render and normalize function.

Note that Live’s normalize always outputs at 0dB, and that there are some technical reasons why this could cause problems (inter-sample modulation distortion if you wanna google it) for some D/A’s. For our uses it’s probably not an issue in this scenario, so feel free to use Live for this if you want. Of course once this is done it’s time to warp the song, and here is the next place we need to pay attention to our levels. I know it sounds like a lot of work, but it’s just something you need to make a habit of paying attention to. It becomes second nature, and these are rules you will use no matter what you are doing with music.

Anyway, when warping songs, I like to also match their levels. That way I don’t have to worry about one track being louder than another when I’m trying to have fun playing. At this point we need to talk about the different types of meters out there. What you see in Live is called a peak level meter. It measures and shows you the highest digital signal that is currently playing on that track. Pure numbers, and the highest one gets display at any given moment. In Live, the highest value seen since you started playback is actually always shown in the oval to the upper left of the meter itself. You can click on this to reset it if you want. This is great, because we can accurately see exactly how close we are getting to 0dB, that ever present no no zone in digital audio.

Except… the human ear doesn’t work that way. Our ears tend to average what we hear when determining how loud something is. So we could be listening to a quiet passage of music with a few brief bursts (only milliseconds long) of noise near 0dB, and we would still think of it as quiet. Our peak meter read those brief bursts near 0dB though, and it thinks the signal is louder than we were able to really hear it as. It’s a difficult concept to grasp, I know, but our ears are incredible good at filtering out and ignoring these short transient bursts. It’s just one example where a peak meter might lead you to believe a signal is very loud, yet your ears tell you it’s kinda quiet.

The solution to this problem is called an RMS meter, which stands for root mean square, a method of averaging values. This type of meter reacts more like the human ear, and can better tell us how comparable two records’ volumes might be. It’s still not totally accurate, I’d say they can get you within a couple dB’s, then you need to use your ears. It takes practice, the more you do it, the better you get. Of course, this raises the point that Live doesn’t have a good RMS meter, so what are we to do?

There’s a couple of solutions available to both Mac and PC users. The first is the free plug in Inspector by RND (formerly Elemental Audio Systems), which you can DL here:

http://www.rogernicholsdigital.com/inspector.htm

The other option is FreeG from Sonalksis, which you can get here:

http://www.sonalksis.com/index.php?section_id=99

Both offer simultaneous Peak and RMS readings, which makes it very easy to get the best of both worlds. Personally, I use Inspector XL as I was able to get it when it first came out for ridiculously cheap, I don’t think I would pay the full price it is now though. The free one works just fine for our uses, as does the Sonalksis one. I also use a different metering scale, the K-14 system. Again, there’s just not enough time for me to explain what the differences are with K-14, but if you’re curious, here’s more info:

http://www.digido.com/modules.php?name=News&file=article&sid=8

Regardless of what plug in you use, the basic principle remains the same. Try and get the RMS (average) level of your tracks the same, while making sure none of the peak readings go over 0dB. To change the volume of track, I adjust via the clip volume fader, as this gets saved as part of the asd file when you warp tunes. Make sure you do this step AFTER you warp your songs. I’ve noticed Complex mode can give some weird spikes on Live’s meters, though I still use it for almost everything.

Typically my tunes end up around being -5 to -6 dB on this fader, though of course it always depends on the particular tune, some can be more, some can be less. Sometimes you need to really trust your ears more than the meters, just use those to get you close. A tune can sound a lot louder if it’s really bassy, and vice versa. Just try and keep the upper peaks of the RMS meter hovering around the same area on the RMS scale no matter what track you’re playing. Ignore the fact that I’m using the K-14 scale, the same ideas apply no matter what RMS meter system you use.

However, keep in mind that I do my mixing with my Allen & Heath DJ mixer, so I don’t have to worry about clipping the master channel. This would be a concern for those who use controllers to mix internally, within the program, and only output a stereo pair. If this is the case, it might be a good idea to play a few tracks together, adjusting them all until they play at the same volume, AND they’re not clipping the master channel.

We’re getting closer to the end of our signal chain when DJing, so it’s time we address the master channel. This is the sum of all the individual tracks you’re playing, and should ideally be left at 0dBFS (FS just stands for Full Scale). To explain why, we need to go back to our bits and decibels again. When you set the bit depth you want to use in Live Preferences, this is also the bit depth of the final output of Live, the digital audio leaving the master channel. We’ve been using 16bits in this guide, so let’s stick with that for another example.

The setting of the master fader determines the final output bit depth resolution (a mouthful, I know) in Live. For instance, say we lower the master fader -6dB to keep from clipping the master peak meters in Live. While we are not clipping anymore, we’ve also lowered the maximum possible bit depth of our audio output by 1 bit. So the best resolution we can ever have now, is 15 bits, even though the final audio file we’d render would be a 16bit file. There would just be a ‘blank’ bit tacked on the end that didn’t represent any of our audio to make it a 16 bit file. I know, it doesn’t sound like a big deal after what we’ve talked about earlier, but it’s worth mentioning. This is the way ALL DAW’s (i.e. Cubase, Protools, Logic, etc) work. It may seem overkill, but I like to keep things sounding as close to the original song as possible, so I’ve always just been in the habit of leaving this at 0dB and adjusting track levels to keep from clipping.

Some people will place limiters on the master channel, so they can run them hotter (louder) without clipping. A limiter is something that basically just prevents audio from EVER going above 0dB, though that comes at the expense of audio quality. Some people also use the Vintage Warmer plug in to do this, as it also add a bit of ‘warmth’ to the audio as it leaves Live. Ironically it does this by mimicking the way analog distorts when you go over 0dB. All this technology just to copy the sound of abusing analog designs! Personally, I don’t use either of these tools when DJing, I prefer to keep the songs as clean as possible. Neither method is ‘right’ though, so use what you like best.

For me, this is largely the last stage of my DJ chain. Like I said, I feed my audio tracks directly to my DJ mixer and mix them with that. Set up as I’ve outlined above, I get signals comparable to what i’d get off vinyl most of the time. If they are a little quiet, I use the channel gains on the mixer. If you’re mixing internally and feeding a DJ mixer or house mixer that way, you can do the same thing, just turn it up on the second mixer to compensate if everything is quieter now. If you’re going directly to an amp, or a stereo system, just turn those up to compensate. This is by far, the best sounding way of doing things IMO.

Hopefully by now you can see why it’s important to know exactly what all those meters are telling you, and why there’s so many of them right front and center in all music apps.

The last place we’ll need to look at metering is when we record our DJ mix and prep it to put on a CD or post as an MP3. For that we’ll need to talk about limiting, compression, and need MUCH more time. So, look forward to another guide from me on those topics in the future.

On a more personal note, if this guide (or any of my other guides) has helped you in your music making, please consider a small $1 donation via paypal to the email address below. Despite an exhaustive job hunt going on for well over a year now, I’m still unemployed and even a dollar here and there really helps me and my family out more than you can realize. Thanks, and I hope you find this guide useful.

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Feel free to pass this document on as you see fit, though I ask that you do not modify it from it’s current form, and give proper credit. If you see any errors, please let me know so I can correct them asap.

Tarekith
Tarekith.com

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