CD mastering information: Letters FAQ

Letters FAQ: Recording tips - peak levels, limiters, headphones, tubes, transistors, dynamic range.

Core FAQ covers analog & digital mastering, DRT's custom gear, sample tracks and pricing.
Background FAQ and Letters offer tips for making better mixes: compression, EQ, CDR issues.
Custom Service is a quick summary for people who target the Major-label market.
Tech News expands on selected technical topics.
One-Minute Tour is where most people start.

Letters FAQ topics:

Here are a few questions received via email:

I'm on a super-tight budget!
Should I spend $$ on tracking, mixing or mastering?

You can't scrimp on any stage without penalty, but if I had a limited budget, I'd make sure that all my tracks were killer onto the multitrack tape. Then if I run out of bucks, I can come back in a few months and mix material that is solid. Same with mastering. If the mixes aren't tight, mastering will help, but only to a point. Get the mixes right first, and avoid finalizing, normalizing or any other quick "mastering" at the mix stage. Finally, when the project sounds tight, master it with someone you trust.

Before you record anywhere, make sure that the engineer you use plays you a tape that he tracked, and that has punchy sounds on it. (Make sure the sounds are still solid when played softly.) If you work with a novice using his home studio, you may get a great rate, but kick and bass sounds will almost certainly not be up to par.

Sorry if I can't give you an easy out. You will hear it when you cut corners, and so will everyone else...

Are CDRs reliable compared to DAT or Internet delivery?
Does lossless data compression help reliability?
Do CD's have more jitter than DAT?

I've certainly experienced manufacturing defects in CDR discs, and I've had trouble reading poor copies from a CDR writer that was dying. But if you use good media with a high-quality writer like the SCSI Plextor units, error rates are often less than 5% of the max allowed by the CD spec. I virtually never get errors writing discs with a setup like this, and I buy CDs hundreds at a time. The discs I produce are checked for errors at CD replicators all over the country, with consistently excellent results. So the error rate of CDRs is not something I worry much about any more. DAT tapes degrade with repeated plays, so I prefer CDR discs. Data compression won't help a file that is stored on bad media. The decompression software will notify you if it can't reconstruct a perfect replica of the original file, but the music is still damaged. Bottom line - use good media.

Another reason that I work with more CDRs than DATs is that many people are interested in 24-bit quality. Standard DATs can't store this, due to their 16-bit limitation. So they burn 24-bit data to CDR and send the WAV files. Works great!

One place where your observations about CDR quality are dead-on is when people burn their audio CDs using Orange-book writers or software. These were designed for track-at-once data CDs, and they often create glitches between tracks. It can be difficult to get a clean transfer in this situation.

Internet delivery of music avoids the errors inherent in both DAT and CDRs. Lossless compressors such as SoftSound or Monkeys Audio can reduce file sizes to 50% or 60% of original, and the error correction is very strong.

Jitter (short definition: distortion introduced by an unstable clock in a digital converter) can only enter the process when you convert between analog and digital domains. Whether you are using DAT or CDR as your final digital medium will make no difference. Using a high quality converter will keep jitter at very low levels.

Is there a standard way to EQ my music?
What are ISO frequencies, and should I use those?

ISO frequencies refer to a series of frequencies used in equalizers and measuring equipment. They are spaced 1/3 of an octave apart, in a series like

20, 25, 31.5, 40, 50, 63, 80, 100, 125, 160, 200, 250, 315, 400, etc. to 20kHz

An equalizer that covers these frequencies between 20 and 20000 Hz would allow you to control audio content anywhere from low bass to high treble with considerable precision.

There are no standard parameters or set guidelines that I know of for equalizing music. You just need to spend time with an equalizer and one instrumental track, and see what you can do to improve the way that the instrument presents itself in the mix. You can learn a lot this way. Naturally, you can try equalizing the entire mix also. Different EQ designs sound different, which is why many people use more than one type for tracking, mixing and mastering. Try checking the Mix Bookshelf at Mix Magazine They may have some book/CD combinations that will be useful.

Is 1/4" analog tape worth the trouble?

If I had a choice between 1/4" analog tape and 24-bit digital at 48kHz or better, I'd probably take the digital. Heresy? Maybe, but there are fine audio cards now for a few hundred $US, that you can throw into a semi-decent computer. They'll turn the machine into a mixdown deck as good as anything you can buy off the shelf. For most material, the warmth/air advantage of 1/4" is not significant, in my opinion, compared to 24-bit. (1/2" tape is another story.) Most of the limits with 24-bit have to do with the recording and mixing technique, not the sound of the medium.

I'll mention another idea, also bordering on heresy: 24-bits can pretty effectively archive the sound of an already-recorded analog tape. For example, you can't _create_ the silky sound of 1/2" analog tape with a 24-bit digital recorder, but if you make a flat transfer off the analog to the digital, the change in signal quality is small enough so that few could detect the difference. I don't know why this is so, but try it if you have a good 24-bit recorder at your disposal. (As a matter of fact, try it with your DAT machine just for fun. You might be surprised at the number of people with good ears who won't hear a difference, especially if 1/4" was the source tape.)

The reason I mention all this is, again, that digital can be sent via Internet, or at least on CDROM discs. You have to be committed to send analog tapes halfway around the world and back... Its fine with me if you want to do it, because we both agree that analog is cool, but consider trying the 24-bit route also. I'm working with clients all over the world now using Internet transfer or CDROM discs, and it turns out fine!

Consider a test master, which will show what incremental improvement I can make. Try sending a 16-bit source - the worst-case scenario - and see what I can do with that. Most tests I do come in on DAT or CDR for convenience reasons. If you like what I do, we can move ahead with higher resolution sources, knowing that matters can only improve.

Tell me more about your monitoring environment.

The dimensions are roughly 26'x12', with diffusion located behind the listening position and a moderate amount of absorption behind the electrostatic speakers, which are positioned approximately 2/3 of the way down the length of the room. The most unconventional feature is that the large electrostatic speakers can swing completely out of the way, revealing the front channels of the 5.1 surround monitor system. This allows the same sweet spot to be used for both monitoring systems, without compromising the performance of either system.

I've generally been influenced by LEDE design techniques since the 1970's, but my room is a bit more live-sounding. Its a deliberate effort to produce "living room" Rt60 times, without serious frequency response or early reflection problems. The tradeoff of this approach is that the speaker positions are critical within a few inches to maintain performance at the calibrated sweet spot. Also, the sweet spot itself is not large. Since I work alone most of the time, this is not a problem.

Another reason that this approach works is that the electrostatic panels function like a line radiator. Their vertical beamwidth is very tight, so early reflections off the ceiling and floor are not much of an issue. ETC analysis confirms this.

I've designed a few control rooms starting with a blank piece of paper, so it was an interesting exercise to see if I could create an accurate environment that did not look as bizarre as many "full-treatment" listening rooms do. I would not advise this approach for others, necessarily. The safest bet is to hire a good designer and let them be responsible for an excellent final result. The do-it-yourself approach works best if you happen to have the training and gear for TEF/FFT/RTA type measurements, and you've studied what other people have done (right and wrong) over the last 50 years.

What about digital noise reduction for tape hiss or mic preamp noise?

Mastering usually makes everything in the mix more clear and audible, including the noise floor of the recording. If you are sensitive to the noise, and have the right processing software, you might noise-reduce the sections of tracks that need it before sending the tracks. That way I can concentrate on bringing out the strong points of quiet source material, rather than fixing problems.

I rarely do noise reduction, because there is a sonic tradeoff between quiet and the smoothness of the top end. Different people have different tastes on this, and since there are so many studios with noise-reduction software available, usually the producer makes that tradeoff under his direct control, and then sends me the final mixes.

The best solution is to not create the noise in the first place. Noise reduction is a band-aid at best. If you have the option to create a mix that is free of noise, that would yield the best final result.

To summarize: I can noise-reduce a section of one or two tunes if necessary. If you need more than that, do it before you send the mixes here, so you can make the tradeoff choices yourself.

Tube vs transistor compression: Does your tube stuff add distortion?

The primary tube processing I use is one section of the Halo processor, which is a compressor and limiter. I can bring it in and out of the signal path with one switch click. It is quite transparent, especially since it can operate in multiple bands. When I first designed Halo, I did a lot of comparative listening to see if it was adding "light but pleasant haze", which is what good tube gear often does. I didn't notice that effect, so I was able to move on to the reason I created it: handling the top 20 dB of the dynamic range.

I think some designers obsess about what's happening 80 dB down. That's not bad, but only if the performance in the top 20 dB is right. The compression action will swamp subtle differences in most cases. The design of the gain control circuit is critical and , with Halo, it was easier to accomplish the task with tubes.

In Rock styles nowadays, levels are being pushed hard, and its not uncommon to have program levels with a crest factor well under 10 dB. It takes some fairly extreme processing to deliver that, and Halo is as transparent as possible in that context. It works great on other program as well. I just finished a Classical choral album yesterday that had tracks with average levels below 40dB. Halo was just trimming the peaks on a few cuts that needed it. This brought a purist recording up to volume levels that would be clear on normal as well as high-end sound systems.

Will a normal Hi-Fi amp work to drive phones?
Should the loudness control be on or off?

A decent home stereo amp (not an all-in-one system or boombox) might do a fine job for the HD600 headphones, depending on it's age and condition. Older transistor amps may have a grittier sound due to component aging and other problems. HD-600's will reproduce this "transistor sound" with great precision, so you may find yourself shopping around for an older amp that sounds good.

As a rule, first try connecting the phones directly to the speaker output terminals. This will give the highest damping factor, which these phones can use to good advantage. You should be careful, though, to check that there are no serious turn-on or turn-off thumps created by the amp. These could toast the phones. Also, be very sure not to do hot-patching or any other risky procedure that could cause a buzz at full volume. That amp has enough drive capability to hurt the phones and your hearing both!

If you don't want to risk connecting to the speaker outputs, and the amplifier has a headphone output, then try that. You will be safer, but the series resistance that reduces the level to the headphone jack may have a sonic effect. Most likely the bass frequencies will not be quite as defined.

If you have access to a low to moderate-power super-quality amplifier at a friend's house, check the phones there just for comparison. You'll find that they reveal differences between amplifiers. Some people use very expensive "tweak-audio" amps with these phones, and love the results.

The Fletcher-Munson equal loudness contour should be used to compensate for the ears' lack of sensitivity to lows and highs when music is played at LOW levels. For normal listening in the 75-85dB range, this contour is just plain wrong. Use it only to check that your consonants are not spitting at you and the low bass was not mixed too hot - a few seconds occasionally. The HD-600 or RS-2 headphones will give you a satisfying sound. Get used to it by listening to a bunch of good commercial music, then switch to your own material.

I'm not surprised if your friends like the mixes with the loudness control set on. They are not in the business of making critical decisions based on what they hear. They want what feels good as they are listening casually. That's not bad, just not the best approach if you want your sound to translate well across different speaker systems. Use speakers to set up imaging, and phones to dial in the tones, at least until you have speakers you really trust. (I have speakers I really trust, and still find that phones are a better reference for certain parts of the mastering process.)

Can you recommend a brand of decent cables for my studio?

Canare, Monster, et al. make interesting stuff. For a short-money option, try MCM Electronics.

RG-59 cable, 75 ohm impedance, gold plated connector with molded strain reliefs:

Part # 32-597 is 10 feet long for $3.29
Part # 32-4505 is 50 ft. at $7.89
Part # 32-595 6 footers are $2.09 in quantities of ten.

These are very low capacitance, and do not use machined castings, which is actually better for skin effect issues. You can spring the shield contacts for super-tight fit. When the solid center conductor finally breaks at the connector from many, many flexings, trash them without a second thought. (Try them as RCA audio patch cords. You may be surprised.)

How should I control dynamics?
Should I dither 32-bit/192kHz tracks to 16/44.1?

I assume you are using software to master your tracks. The Waves L1 limiter is commonly used to bring up softer passages. Sound Forge has a limiter that will do this also. Magneto and other software will do this while trying to create an "analog-like" soft limit. I prefer the sound of analog compression and limiting, which is what I use.

If you plan to master at at an outside facility, I would dither to 24bit/96kHz, which is a much more prevalent standard for interchanging audio. Actually, I have many masters come in at 24-bit/48kHz, which is also an even division of 192K. This can sound quite good if the original recordings are in good shape, and it works for people who have limited disk space.

Dithering from 192 to 44.1 is a hairy mathematical formula, and many pros prefer the sound of a good analog transfer to the sound of the software equivalent.

We're in NYC. Can we attend the session?

I'm not doing attended sessions at this time, since I have a 4+ week backlog pretty consistently. My work is guaranteed, and I've found that I can listen to source tracks, then cover all the points necessary via phone. This formula has worked well for years. As an aside, I find, like many other mastering engineers, that it is easier to get into a groove and maintain concentration when working alone. This results in a better album.

If you have mastering decisions that you need to make personally on the spot, I would strongly advise working with a local facility. As a matter of principle, working locally is usually better. My business exists mostly to serve folks that cannot find - locally - both the human factors and technical resources that I offer. (Somewhat to my surprise, I do quite a lot of business with NYC people who have mastering facilities practically in their back yards.)

Why is it good for program peaks to register above -6db on the meters?

Each bit of a 16 bit recording gives you 6dB of dynamic range, for a theoretical total of 96dB. If your recording never peaks above the -6dB level, you are essentially "throwing away" one bit of potential dynamic range. Then you have a 15 bit recording, since, unlike analog, there is a hard limit to how small of a signal can be stored. Once you get to -96, the signal is gone. (This assumes that you are not using noise-shaping or other techniques that work to increase apparent dynamic range...)

In the real world, even if your levels peak at zero, the noise floor of your console and the recording space itself will probably limit dynamic range to 70 or 80 dB. The upside of this situation is that the small level of hiss and hum works like a dither signal, and the human ear is remarkably good at picking signals out of a noise floor. (Think of following a conversation at a loud party. Microphones fail miserably, but human ears do it easily...) Summary: your console is providing dither noise, which can help out with resolution of low level signals. Still, make use of the 16 bits you have. If your mixes peak between -6 and -1 on a digital meter, you should be fine. You might also consider upgrading to a 24 bit mixdown deck.

Is internship a good way to learn recording?

Internships are a proven way to break into recording. Internships are more common at higher end studios in any given area, since they are busy and have the greatest staffing needs. The process has been compared with indentured servitude. Even if you come out of a known academic recording program, you will often start off maintaining session documentation, learning and performing the favorite mic setups for senior engineers, and running for pizza. After some time, you can contribute as an assistant on large sessions, and you can run the smaller ones. Aligning analog tape machines is part of the curriculum, as is the human side - dealing with lack of sleep, cranky producers and clients. Finally, management decides if they want to keep you, or if its time for you to move on. You need aptitude, persistence and positive attitude.

Sennheiser HD600 phones: For Classical music only?
Can they play loud?

Regardless of specs, the 600's just sound right. The Stax Lambda Nova's that I recommend even more highly are virtually the only phones that Classical editors use if they have a choice. The reason is flat response, very low distortion, high resolution etc., but basically, you can hear everything better.

If you are pushing any of these phones to SPL levels where you approach the impact SPL limit of the transducer, you are monitoring too loud to preserve your hearing over the long term. Very few pros monitor for long above a maximum of 85dB. If they do, listening fatigue sets in and judgement gets cloudy. If your transducers have enough resolution, you don't need to push levels.

I need to remaster an analog tape from 1980.
Can this gum up the heads on my tape deck?

Play fifteen seconds of the tape, then check the heads. If they are completely clean, play another minute. If still clean, try three more minutes. If heads are still clean, you do not have the "sticky shed" problem that will gum up the heads.

If the heads are gummed up, you will also notice that the treble in the music has been reduced. Stop before you damage the tape further. Clean the heads. The solution that I've used is to bake the reel of tape for five hours at 140 degrees Fahrenheit in a convection oven. This will evaporate the water that has penetrated and partially dissolved the binder. This binder holds the oxide to the plastic backing of the tape. Convection vegetable driers work for this purpose, as will a normal oven, TURNED OFF, with a 120 watt light bulb in it.

If you use this method, get a cooking thermometer and plan on spending three hours adjusting the oven to the right temperature. (Crack the door open if necessary, put a little fan in there to circulate air and even out temperature variations. Try a different size bulb. Fiddle around as long as you have to to get the temp right.) Five degrees either way is not a big deal. Fifteen degrees hotter will work faster but might stress the tape. After the bake, let the tape cool slowly to room temperature before playing. A tape treated this way will probably be good for a few months before it begins to absorb moisture from the air again.

Archive the restored tape to digital if you can. Use the highest resolution recorder that your local studio has. Then use the restored tape as you normally would.

Any other ideas for improving my recordings?

Here are some things to look at:

Arrangement. Build tension. Can you think up three rhythmn keyboard/guitar parts that would be just killer? Cool. Now only use the tastiest of those three - very sparingly - after the first verse has established the bare-bones killer groove of the song. Now you say "But the parts are all great." I know, but keep it simple. If you just have to use them all, use them on the tail of the tune during the final jam before the fadeout. This will build tension so that people listening again (and again) will feel anticipation.
Don't try to be a one-man army. Find a collaborator. If you don't know anybody, go to the nearest instrument store and ask the salesman who plays and records locally. Pay them 20 bucks to come over and comment on your effort. They will be pleased you asked, and you may pick up tips and maybe even a player for your sessions.
Its tough to beat a P-Bass with round wound strings. Try one direct box, one mike on the cabinet. You want a definite attack, with a round tone. Watch out for sub-bass tones below 40 hZ. They eat up airspace. The exception is 5 string bass or HipHop low tones. Even in those cases, most bass comes across in the second harmonic. Thats why the 100Hz band has worked as a means of controlling bass EQ for the last 50 years. Make sure each bass note has an audible attack, locked to the kick drum when the kick plays on the same beat. Bass with no attack fills up the low end, but it does not help the drive of the tune.
Do the housekeeping! Solo the individual tracks, and erase or mute any studio chatter, coughs, tune-ups or chair squeaks that might be hiding there. (Yes, I've been guilty of ignoring this rule when pressed for time during mixdown. As a result, one example - that I am responsible for - now sits on an album by a multi-platinum artist. The fact that it is a minor oversight is no excuse. I won't forget the lesson.)

I'm in school and want to learn more about recording. Suggestions?

Check out magazines such as Mix, Electronic Musician, EQ, Recording etc. They offer free subscriptions and have websites as well.

Many people pick up a used recorder, a couple of mics and just dive in. Offer to record you friends' bands for free. If you have a studio nearby, offer to help out for free. (Even graduates of recording programs often have to work for little or nothing for a while.)

I started out playing drums in a Rock band at 15. We built our PA speakers and dragged our stuff around in my parent's station wagon. That's how I started with audio. Before that I built radios. I found that almost everything technical that I learned came in handy later.

Shop eBay or a used book store for a copy of Tremaine's AudioCyclopedia. I have a copy from the early '70s that can teach you a lot about recording techniques, equipment etc. Also check out sites like Home Recording.com

Almost all the "recording studio in a box" processors like the Roland VS880 and 1680, and the clones from Yamaha etc, have mailing list groups where people hang out on the web and trade tips and questions. Go to Deja.com and search. The newsgroup rec.audio.pro has a huge archive you can look through to answer questions.

What about analog tape emulators and 3D effect generators?

A lot of plugins and hardware digital devices work to create a "virtual analog" sound in the digital domain. I've tried units ranging in price into the multi-thousand $$ range. They're fun, but they can't match the real thing. That's why people still use 2" multitrack, 1/2" analog mixdown decks, and analog mastering gear. Same with the 3D/ambiance processors. I've gotten plenty of email from people who use digital versions, but they are still looking for a quality of air and space that analog gear does better. It will be cool when you can convincingly do it all with a computer, but the technology is not there yet.

Core FAQ covers analog and digital mastering, DRT's custom gear, sample tracks and pricing.
Background FAQ and Letters offer tips for making better mixes: compression, EQ, CDR issues.
Custom Service is a quick summary for people who target the Major-label market.
Tech News expands on selected technical topics.
One-Minute Tour is where most people start.

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