What is Dynamic Range and Why is it Important?

You may have heard about or read some articles regarding the "Loudness War" or the loss of dynamic range in modern music. I'm about to school you on this terrible trend in modern music and how it makes your music weaker, even if it is louder than the next guy.

Simply put. In audio the dynamic range is the difference between the average and peak levels of a signal. An audio file that has an average level of -14 dB RMS and a peak at -0.1 dBfs is said to have a dynamic range of 13.99 dB.

I often get the request to make a record "loud, yet punchy" which I'm afraid to inform you is an oxymoron and you can't physically have both.

Imagine in your head the image of a speaker and how it moves in and out while music is blasting through it. How far does it move out when you crank the volume on a record you REALLY love? Every time the kick hits that speaker moves out very far and very fast. It moves a lot of air very quickly and this is where we perceive the "punch" of a kick or other percussive elements in a song. The harder and faster that burst of air jumps out above the average signal, the more impact it has on us.

What happens when we raise the average level of a signal with heavy compression?
The further we raise the average level the less those percussive elements are able to jump out. Get the speaker picture in your head again where before the speaker was moving in and out about an inch on average but would push out 3 or 4 inches when the kick hits. Now let's raise the average level where the speaker is staying extended between 2-3 inches most of the time but still only pushes out to 3 or 4 inches to fully extend when the kick hits. The speaker is loud but it can't physically move that big sudden burst of air. Only a tiny bit of extra air is now being pushed when the kick hits in relation to the average signal.

Does it all make sense yet? Not only does hyper compressing your music in mastering make your dynamics weak, it also adds a TON of distortion in the process. Now that you know the physics behind it all let's listen to some examples.

In the following example I have a simple TS909 drum loop. The first 4 measures are uneffected. The next 4 measures following have been ran through a limiter to give the loop only about 6 dB of dynamic range which is currently the norm on hyper compressed tracks. The last 4 measures is for those of you who really like the tone and nastiness of the compressed track but like the punch and dynamics of the uneffected track. The technique of blending a compressed track in with an uneffected signal is known as parallel compression aka the New York compression trick. All have been volume matched to take loudness out of the equation.

Download the wav file here, burn it to a CD and audition it on a good set of speakers capable of reproducing a decent amount of bass.

When used correctly compression can add some nice color to the original signal. On the right system the compressed track will punch you even harder than the original. When compression is abused, especially the type of hyper limiting used in mastering, you end up with a weak and distorted signal that doesn't push as much air and doesn't have anywhere near the same impact.

If you like distortion as an effect put it in parallel with the original track so it not only sounds dirty, but retains the punch of the original.

If you have a dynamic record that has a lot of impact your fans will reach for the volume control and turn it way up. Enjoying it all the way through. If your album isn't that dynamic and has a lot of distortion from limiting your fans will likely get fatigued quickly, turn the track down, and only listen to a track or two instead of your entire album.

Bob Dylan was famously quoted as saying "You listen to these modern records, they're atrocious, they have sound all over them," he added. "There's no definition of nothing, no vocal, no nothing, just like ... static". Part of what he is referring to is this use of modern processing to destroy any natural dynamics in recorded music.

Loudness is a totally relative thing. Your fans have a volume control and won't hesitate to use it depending on how your music makes them feel. It all depends on if you're willing to be a solution or a part of an ongoing problem. I certainly have been a part of the problem myself by going there when a client requests. But if given the personal choice between have a smashed and loud record or a dynamic open one I'll choose the dynamic record every single time.

For more info check out
What Happens to My Recording When it’s Played on the Radio?
Dynamic range - Wikipedia
Loudness war - Wikipedia

Getting the Best Levels While Tracking

The first thing you read in most manufacturers manuals or recording books about setting levels is to "get the level as hot as possible without peaking". What they don't explain is how this leaves you with exactly zero headroom on the mix bus when it comes time to mix the track.

Sometimes it reads to get as close to 0 dB without peaking without specifying what type of dB. In a lot of recording books this means 0 dBu which is COMPLETELY DIFFERENT than 0 dBfs (full scale).

The first is from a school of thought that comes mostly from the 16-bit digital recording era where "using all the bits" was important in insuring a low noise floor. With today's 24-bit and higher signal paths this is no longer advisable and in fact detrimental to proper recordings.

If you have already tracked your project this way the best advice I can give you is to use a trim knob or gain plugin on each track to turn the levels down by 10 dB and possibly even further than that. You want to turn everything down equally until your loudest peak with every track playing is no longer clipping the master bus. Unfortunately you may have already gotten some unwanted distortion from your analog to digital convertor and there is no way to remove it except to start over and retrack everything.

There is a better and faster way to set levels while tracking. But it requires some research on your part to find out exactly how your recording hardware was calibrated.

Bust out the manual that came with your audio interface and find the page that lists all of the technical specifications for that device. What we're looking for is a maximum or full scale input value that is listed as + dBu. I will list the values of a couple of different popular recording interfaces here so you can see exactly how varied these measurements are.

Avid / DigiDesign Mbox2: Maximum Input +21 dBu
PreSonus FireStudio: Maximum Input Level (Unity Gain, 1KHz @ 0.5% THD+N) +17 dBu
TC Electronic Konnekt: Full Scale Input Level @ 0 dBFS +13 dBu

The difference in the maximum input level between manufacturers is relatively meaningless and doesn't really mean that one device has more or less headroom than the other. What it does tell you is what the particular designers of the hardware had in mind in reference to the calibration of the analog to digital convertor.

How do I use this value to get a better recording?
What the designers are telling you with this value is that the nominal input value is 0 dBu and their device is designed to work best at this level. The +dBu refers to exactly how much headroom you have before you totally clip and distort the signal. But, it still works best at the 0 dBu point.

0 dBu is a holdout from the analog recording days and most audio interfaces are still designed to interface with older outboard gear from this era. This was an era before instantaneous digital peak metering was the norm and the VU meter was king.

The VU meter has a slightly softer ballistic than a digital peak meter. It was designed to "look good" when you ran a voice through it. It was not very accurate but nonetheless became the standard audio meter on every console, tape machine, and radio.

What we need to get an optimal level is a VU style meter in the box! I recommend Sonalksis FreeG because it's free, it's multi platform and you can make it work in just about every single DAW out there.

Before we track we need to load up an instance of FreeG in our channel strip. Click on the Sonalksis logo and change the ballistics style to VU.
Now we're ready to set our level! Remember the maximum input value of your interface? Subtract that from 0 dBfs and that becomes your new 0 dBu point. So if you had a maximum input of +13 dBu you would slowly turn up the gain until it peaked at -13 dB on the FreeG meter in VU mode.

This is your optimal tracking level. This is how you get the cleanest crystal clear audio possible into your computer.

Keep in mind that you still need to watch the digital peak meter to make absolutely certain that you don't hit 0 dBfs but for most sources you'll never end up near that level and when you go to mix chances are you'll still have enough headroom left over to comfortably start your mix without distortion.

You whine that it's not loud enough... TURN THE VOLUME UP ON YOUR SPEAKERS DUMMY! Be sure to read my article hereon how to get the most out of your studio monitors and this will all start to make sense sooner or later.

Easy Tips to Get the Most Out of Your Studio Monitors

Probably the most overlooked aspect of home studios are the acoustics of the control room. You may put money into great instruments, mics, and preamps, but if you can't hear what's going on in your monitors how do you expect to make a great record?

When people ask me for tips on mixing music I tell them that if they can hear what they are doing accurately it should be easy. What I mean is, if you have been listening and playing music for a good part of your life you already have a great benchmark for what good music sounds like, right?

We need to be able to accurately hear what is going on in our mix or else it will only sound good on your own set of monitors and that's no good. There are a few hard and fast rules worked out by engineers much smarter than all of us when it comes to acoustics. The least we can do is honor their work. So here are a few easy tips that will help almost any set of studio monitors and get your mixes to that next level.

Rectangular rooms are more ideal acoustically. If you can, try to move your mix out of a square room into a rectangular room. The bigger the better. If you can move from a rectangular room into a large oval room... wow, that's a great room.

The best spot in the room? 38% back from the front wall. The second best spot in the room? 38% forward from the back wall. The worst spot in the room? Right in the center. Don't ask me why. This isn't the place for why. I'm telling you how. If you want to know why just use Google ya' dingus.


The general rule of thumb is to have the tweeter at ear height if it is a two-way speaker or to have your ear sit right in the center between the tweeter and mid range driver if it is a three-way speaker.

While the powers that be may disagree on the exact angle they all agree on one thing. Your sweet spot is part of an equilateral triangle with the speakers i.e. both speakers should be exactly as far from you as they are apart. You'll also want to keep them away from the walls. Also be sure to fire them into the long part of the room.

Dolby Placement Specs
DTS Placement Specs
THX Placement Specs

One of the cheapest tools for audio professionals is an SPL meter. Not only is it handy for checking a level to make sure you don't blow an expensive ribbon mic. It can also be used for setting up your speakers! You can purchase one from any RadioShack location (Catalog# 30-2055) or buy one online here.

This is very easy to do and I will provide you with the test tone to do it here. Download Pink Noise Test Tone

Load the test tone up in your DAW on a mono channel with absolutely no effects. Leave the fader at unity. Don't have a single effect on your master bus. It's at a very specific level for a reason. That's why it's called a test tone. DON'T MESS WITH IT DUMMY!

Get your SPL meter out and set it to C weighting with a slow response. Set the meter up to be sensitive to around 80 dB. If you have a camera tripod handy you can affix the meter to it and set it in the sweet spot. If you don't have one available get a friend to hold the meter in the sweet spot where you head would be.

Pan the test tone hard left and play the tone. Adjust the volume on your left speaker up or down until it reads exactly 83 dB SPL on the meter. KEEP THE METER IN THE SAME PLACE! Now go and pan the test tone hard right in your DAW and play again. Adjust the volume on the right speaker until it reads exactly 83 dB SPL on the meter. If you did everything right when you pan back to center you you get a readout of 85 dB SPL on your meter.

If your speakers don't have individual volume controls this same thing can be achieved with a volume and pan control on your receiver.

Congratulations! You have now properly set your speakers to an industry standard reference level. The trick with a reference level is to let your speakers work for you by telling you when something really is too loud or too soft. Don't touch the volume on your speakers or your master bus while mixing. If it sounds too quiet, turn up the faders. If it sounds too loud, turn them down. Easy.

If you have a subwoofer in your setup you can also calibrate it the exact same way. Just make sure the test signal is only running to the subwoofer when you set it's level.

Mixing with a reference level lets your focus shift to the music instead of watching meters all the time for clipping.

We all love to listen to music loudly at home, in our cars, and at the club. But when you listen too loudly while mixing there are a couple of things that happen. Not only will long term exposure to loud levels while mixing will destroy your hearing, loud levels will also mess with your perception of low and high frequencies. Enter the science of equal-loudness contours.

The science of equal-loudness contours basically states if we listen at too low a level we won't hear enough bass and if we listen at too high of a level we won't hear the correct level of high frequencies. Turn your SPL meter back on with C weighting and a slow response and measure the level of your mix. If it's coming in somewhere between 77 dB SPL and 85 dB SPL you're in the right loudness range to hear what you need to be hearing most of the time. Of course you can always push your volume up or down occasionally to check and make sure your mix doesn't fall completely apart at different levels, but if you stay in this particular window you'll find your mixes get the balance they need a lot faster.

You can find more info on equal-loudness contours here.

Headphones can be great for EQ'ing certain problem spots in a mix because they take the non linearities of the room out of the equation giving them a flatter frequency response in most cases. They do have one MAJOR drawback and that is the omission of a phantom center channel. When this is missing it makes it very hard to properly set a level for the crucial mix elements that live there i.e. kick, snare, bass, vocals.

When you listen to a mix on a set of speakers that have been properly calibrated the elements in the middle sound about 2-3 dB louder than the same mix does in headphones. Not only that but there is a very defined placement of the center elements on a set of speakers whereas the phantom center in a set of headphones just comes from some random place in your head.

If you absolutely must use headphones to mix I like to suggest a product like 112 dB Redline Monitor which emulates a set of speakers in your headphones. It's not as good as speakers, but it's better than just the cans.

Even the best studios in the world have several sets of monitors to check their mixes on.

One easy and accessible extra set of speakers to check your mix on are the cheap white earbuds that come with Apple iPods. They are terrible and hard to do an entire mix on but if anything sounds extremely whacky in them it's worth trying to fix it because a lot of people will listen to your mix on this exact setup.

I also recommend checking your mix on every system that you're familiar with. Your own car. Your TV/home theater setup. Your clock radio.

Where not to check it would be systems you aren't familiar with. Your friend's car, etc. If you don't listen to the system every day you should not make ANY mix decisions at all based on what you heard from those systems.

In conclusion there is a lot that you can do with your monitoring setup that is free or that costs very little to improve your sound dramatically. Yes, acoustic paneling is, in my opinion, a must in every professional studio. But if you are only putting out records very infrequently or just doing this as a hobby you may not be willing to put the money it takes into proper acoustic treatment. But if you follow these tips you know you will be getting the best sound that you can get without spending that extra money.

If you have any questions, comments, or additions please be sure to let us know below.

Ensuring Your Disc is Ready for Manufacturing

Something I run into a lot these days are customers who just want their files processed by me but want to handle editing, sequencing, and disc burning themselves. This is something that may seem easy and straightforward until you send your disc off for replication and the plant rejects your homemade disc.

There are several reasons why your disc may be rejected and this varies among manufacturers. I will cover a few of them here and give you a few pointers on how you can ensure that your audio will transfer from your DAW to the final product with as few technical issues as possible.

The first and certainly the most common reason that discs get rejected at the plant are because they have a large error rate which in turn causes the disc to drift out of Red Book specs. These errors can be caused by several different things including cheap blank media, cheap burners, and vibrations while burning the disc. Other reasons a disc might be rejected could be CD-Text with unrecognizable characters, and other incompatible or incomplete subcode data.

"What steps can I take to make sure my disc doesn't have any errors?"

1. DON'T TOUCH THE DRIVE OR MOVE/BUMP YOUR COMPUTER WHILE BURNING! If using an external drive be sure to isolate it as much as you possibly can. Foam Auralex type pads that are used to isolate studio monitors are great for this purpose.

2. MAKE SURE YOU USE MEDIA OF EXCELLENT QUALITY! It's widely accepted that media currently manufactured by Taiyo Yuden is among the best out there. I recommend their Watershield products. The error rates reported are very low especially when coupled with and excellent quality CD burner.

3. BURN AT A LOWER SPEED! The ideal speed to reduce errors really depends on the media/burner configuration you are using but speeds between 4x and 16x are typically used. Just because your media has a maximum speed rating of 52x does not mean that it will perform it's best at 52x. What works best is really hard to gauge without any type of in house error checking but burning at a lower speed definitely seems to help.

4. USE A GREAT BURNER! If you have the spare cash lying around it's a good investment to put some money toward a burner that is dedicated to burning audio CDs. Collected data shows that most CD/DVD/Blu-Ray combo drives report higher errors than those drives that only operate on CD media. A lot of mastering houses swear by Plextor Premium and Premium 2 drives for their reliability and error checking capabilities. Pioneer is also known for quality drives. Read up on what current production models are recommended by searching at the Hydrogen Audio and Gearslutz forums.

The Plextor Premium drives ship with software called PlexTools that lets you check for the same errors a CD replicator checks for before they approve your disc. If you are putting out releases on a regular basis, having one of these drives available for error checking is an invaluable tool that will save you a lot of time.

Understanding the C1/C2/CU errors reported by PlexTools is fairly easy. C1 errors (also known as the BLER rate) are correctable errors and 100% of the discs you burn will have some amount of C1 errors. The Red Book specifications allows for a 220 error per second averaged over any 10 second period. This is, in my opinion, way too high. But it gives you an idea of what is playable without a lot of noticeable degradation of the source material. C2 errors, while correctable in most modern players, are definitely undesirable and I will burn the disc again if it contains even a single C2 error. CU errors are the worst kind. Completely uncorrectable and a disc with even one will most certainly be rejected by the manufacturing plant. If your disc has even one single CU error you need to burn your disc again.

Lower values are always preferred, but this disc is well within spec and will have no problems producing a high quality glass master.

Is there a better way available to transfer my disc to the plant?

Why yes, Billy, there is. There is a 100% infallible method of transferring your disc to the plant digitally without ever burning your project to a disc and that method is known as a DDP (Disc Description Protocol) image. A DDP image is similar to that of an ISO disc image with built in error checking so if there is some problem in the transfer and it arrives to the plant without being 100% bit for bit identical to the original it will be flagged and you will have another chance to transfer the image.

Of course, you need to have software that is capable of exporting your final sequence as a DDP image. Fortunately this is no longer as cost prohibitive as it once was. If you have Apple Logic installed you most certainly have WaveBurner installed along with it which is great for final sequencing and it can export a totally compliant DDP image. Standalone solutions for both PC and Mac include DDP Creator by Sonoris Audio Engineering.

In conclusion, it's because of these reasons that I really push to do the final editing and sequencing for my customers. In addition to the above there is also the possibility that bad edits can leave your final product full of pops and clicks at the edit points. I've never charged any extra for this service and have always factored it into my rates but music is near and dear to it's creator and I understand the need to exercise as much control over the final presentation as possible. At the very least I recommend letting your mastering engineer check your final edit for technical errors before it's sent off.

If you have any questions, comments, or additions please be sure to let us know below. Until then, keep making that excellent music.

Selected Discography

Audio Examples

One thing to keep in mind while listening is to remember that every project shares a different vision and what applies to one mix will almost certainly NOT apply to another. Enjoy.

Here is a fantastic song by Look Afraid with a great mix by Mitch Wyatt of Sounds Underground.



I love this clip because the changes are very subtle and tasteful. Everything good mastering is to me. It is also extremely dynamic for the volume level we achieved.

Mastering Q&A!

Q: What sources do you accept?
A: We accept AIFF, APE, FLAC, SD2, and WAVE files, as well as audio CD. You can mail them to us on CD-R/DVD-R, hard drive, or upload them to our FTP server.

Q: Can you master from an analog source?
A: No, we only master from digital sources. This ensures the upkeep costs on our equipment stays low. You can, however, have your analog source captured digitally on your own converter and sent to us. We recommend doing this at 24-bit 192 kHz to preserve all of the glorious analog details.

Q: What bit depth and sample rate should my files be in before I send them to you.
A: Simple answer, the exact quality you recorded them in. The higher the better. Please do not convert bit depth or sample rate before you send them to us. We have high quality converters that sound amazing specifically for this purpose.

Q: Can I use a compressor on my master bus?
A: Unless it's applied very conservatively for glue/color we don't recommend it.

Q: Can I use an EQ on my master bus?
A: We don't recommend it.

Q: How much headroom should I leave on my mix?
A: We prefer mixes come in between -20 dB RMS and -14 dB RMS while making sure you never clip the master bus. Miracles have been performed with far less headroom, but again we don't recommend it.

Q: Should I normalize my files before I send them?
A: Never.

Q: Should I fade-in and fade-out my mixes before bringing them to be mastered?
A: This is entirely up to you. We often apply fades during the editing process. If you apply fade-ins and fade-outs during mixing, we generally leave those fades intact. However, just like master bus compression, it is sometimes difficult to modify an existing fade. If you choose to apply your own fades, we recommend that you also provide an additional version without the fades.

Q: Do you use the Waves plugins?
A: This is where I lose work, ha. I use exactly one plugin made by Waves and that is the Renaissance Compressor. Regardless of what marketing hype and the retail price of the Waves plugs says to you they are, in my opinion, not suitable for mastering. There are many FREE and nearly free plugins out there doing a better job than what's in the Waves mastering bundle.

Q: But the Waves plugins are the best and cost $10,000!
A: No, they're not. Seriously. The sooner we all stop judging studios and engineers by the plugins they use we'll all be better for it.

Q: Why are your rates so low?
A: Several factors contribute to this. For starters, Neon is based out of a (mostly) digital home studio. This keeps business overhead and equipment upkeep costs to the bare minimum. Finally, we are firm believers in DIY ethics and believe everyone should have the means to put out a great sounding record, not just record labels.