Mastering is an art and a science.

Mastering is the final creative and technical step prior to pressing a record album (CD, DVD, cassette, or other medium). Compare mastering to the editor's job of taking a raw manuscript and turning it into a book. The book editor must understand syntax, grammar, organization and writing style, as well as know the arcane techniques of binding, color separation, printing presses and the like. Likewise, the Mastering engineer marries the art of music with the science of sound.

The Craft of Mastering

The audio mastering engineer is a specialist who spends his or her entire time perfecting the craft of mastering. Audio mastering is performed in a dedicated studio with quiet, calibrated acoustics, and set of wide-range monitors. Signal paths are kept to a minimum and often customized gear and specialized tools are used. The monitors should not be encumbered by the interfering acoustics of large recording consoles, racks or outboard gear. In other words, the acoustics are first optimized, and all other considerations must be secondary to the acoustics. For optimum results, mastering should not be performed in the same studio as the recording or with the same engineer who recorded the work. It is important to find a mastering engineer who will bring his expertise and unique perspective to an album project, to produce that final polish that distinguishes an ordinary recording from a work of art.

What Is a Mastering Engineer?

The mastering engineer must have a musical as well as technical background, good ears, great equipment, and technical knowledge. Ideally, he should know how to read music, and have an excellent sense of pitch. He knows how to operate a range of specialized technical equipment, much of which is not found in the average recording studio. The successful mastering engineer understands many musical styles (and there are a lot out there!), edits music, and puts it all together with sophisticated digital processing tools. He is sensitive to the needs of the producer and the artist(s), and treats each project with individual attention. He must understand what will happen to the recording when it hits the radio, the car, the internet, or the home stereo system.

What's the Difference between a CDR and the Glass Master?

Premastering, not mastering, is the more accurate term, since the true master for a Compact Disc is called the Glass master, which is etched on a laser cutter at the pressing plant. In fact, the Glass Master is destroyed during the production process. The only thing permanent is the stamper, a round metal form that can be used to press thousands of CDs before it is replaced. There are two intermediate steps (the father and the mother) before creating the stampers that press your CDs. If you're interested in learning more about the processes at the plant, visit So, we really should label the material that is going to the plant a PreMaster. The master for a compact disc must be in one of two forms: a DDP file, or an audio CDR (recordable CD). Both of which contain exactly the audio which is to be replicated as well as the PQ (track) data and possibly CD text or some graphics. Even though it's really a premaster, it's customary to label it a CD Master--because (hopefully) there will be no further alteration of the digital audio at any subsequent stages. If the pressing plant does its job right, the bits on the final CD will be identical to those on the master that left the Mastering House. If you're interested in learning more about how CDs are made, take Cinram's virtual plant tour.

Why shouldn't I call my original file the "MASTER?"

The word Master is overused. I've searched record company libraries, and often found several tapes of a record album, each one labelled master, but in reality, there can be only one Master. You should label your tape or file Final Mix, or Original Session or Edited Work Parts, or Edited Compilation, Unlevelled or perhaps Assembled Submaster. But as you can see, using the label Master will only confuse things later on. Other confusions arise when the producer has second thoughts. He may decide to change the EQ or relevel a song, but forget to relabel the previous master. Certainly, the first thing is to prominently print DNU (Do Not Use) on the label of a newly obsolete tape.

Can't I just mix and put the file on the Internet or CD? Seven reasons why you need mastering.

Every recording deserves good mastering. When you're through mixing, your work is not finished. Mastering adds polish, it sounds more than just a record...it becomes a work of art. The songs work together seamlessly, their sound can take on a dimensionality and life that enhances even the best mixes. Here are seven reasons why Mastering is needed.

1. Ear Fatigue
Most music today is produced by recording multiple tracks. The next step is the mixdown. This mixdown may take anywhere from 4 hours to 4 weeks, depending on the producer's predilections, the artist's whims, and the budget. Usually each tune is mixed in isolation. Rarely do you have the luxury to switch and compare the songs as you mix. Some mixes may be done at 2 o'clock in the morning, when ears are fatigued, and some at 12 noon, when ears are fresh. The result: Every mix sounds different, every tune has a different response curve.

2. The Skew of the Monitors
Monitoring speakers. It's amazing when you think about it, but very few studios have accurate monitor systems. Did you know, placing speakers on top of a console creates serious frequency response peaks and dips? A typical control room is so filled with equipment that there's no room to place a monitor system without causing comb-filtering due to acoustic reflections. And though your heart is filled with good intentions, how often do you have time to take your rough mixes around, playing them on systems ranging from boomboxes to cars to audiophile systems? Usually there is no time to see how your music will sound on various systems in different acoustic environments. The result: your mixes are compromised. Some frequencies stand out too much, and others too little.

3. More Me
The producer was supposed to be in charge. He tried to keep the artists out of the mix room. But something went out of control. The producer was gone for the day, or the bassist had a fit of megalomania. Or the artist decided to be his/her own producer. Whatever....all the mixes sound like vocal, or bass, or (fill in appropriate instrument) solos.

4. May I Have Your Order, Please
When mixing, you (the producer) often have no idea what order to put the tunes until after all the mixes are completed. If you physically compile these songs at unity gain, and listen to them one after another, it probably won't sound like "a record." Some tunes will jump out at you, others will be too weak; you may discover (belatedly) that some tunes are too bright or weak in the bass, or that the vocal is a little weak, or that the stereo separation is too narrow. These things actually happen, even after weeks in the studio, and the problems sometimes don't become apparent until the album is assembled in its intended order, or auditioned in a good monitoring environment.

5. The Perspective of another Trained Ear. The Buck Stops Here.
The Mastering engineer is the last ear on your music project. He can be an artistic, musical, and technical sounding board for your ideas. Take advantage of his special ear... many beautiful music projects have passed through his studio. You may ask him how he feels about the order of your songs, how they should be spaced, and whether there's anything special that can make them stand out. He'll listen closely to every aspect of your album and may provide suggestions if you're looking for them.

6. Midi Madness
Lately it sounds like everyone is using the same samples! Acoustic sounds are coming back in vogue, but perhaps you haven't got the budget to hire the London Symphony. So, you had to compromise by using some samples. But you shouldn't compromise on mastering. Good mastering can bring out the acoustic quality in your samples, increasing your chance of success in a crowded music field.

7. Don't Try This at Home
The invention of the Digital Audio Workstation (DAW) and the digital mixer is an apparent blessing but really a curse. Many musicians and studios have purchased low cost DAWs and digital mixers because they have been led to believe that sound quality will improve. Unfortunately, it's real easy to misuse this equipment. We've found many DAWs and digital mixers that deteriorate the sound of music, shrink the stereo image and soundstage, and distort the audio. There are several technical reasons for these problems-usually wordlength and jitter are compromised in these low-cost systems. Therefore, we recommend that you protect your audio from damage; use a mastering studio that employs a high-resolution system that enhances rather than deteriorates audio quality. Prepare your tapes properly, and avoid the digital pitfalls. Use the informative articles at the Digital Domain web site as resources to help you avoid audio degradation. When in doubt, take this advice: mix via analog console to a high-resolution file or to analog tape, and send the original tapes or files to the mastering house. You'll be glad you did. Those are only some of the reasons why, inevitably, further mastering work is needed to turn your songs into a master, including: adjusting the levels, spacing the tunes, fine-tuning the fadeouts and fadeins, removing noises, replacing musical mistakes by combining takes (common in direct-to-two track work), equalizing songs to make them brighter or darker, bringing out instruments that (in retrospect) did not seem to come out properly in the mix. Now, take a deep breath and welcome to the world of mastering.

Analog vs. Digital Processing in Mastering

Earlier in this article, I cautioned against returning to the analog domain once you've converted to digital. Ideally, you only want one of these conversions, once in the original recording, and once in the CD player playback.
But what about Pultecs, tube and solid state equalizers, tube and solid state compressors, limiters, exciters... Most mixing engineers can cite a plethora of famous processors that perform their work with analog circuitry. While useful for effects patching during a mixdown, a good number of these processors are unsuitable for mastering purposes. For example, an old, unmaintained Pultec may be a little noisy, but still be suitable to process a vocal or instrument during a mixdown. But would you pass your whole mix through that noisy box (maybe yes, if you like the sound!)? However, every processor used by a mastering studio (a good mastering studio) will be used in matched pairs, have calibrated positions, be quiet, clean, well-maintained. Calibrated positions are important for re-mastering, or for revisions. Clean means low-distortion and noise. Matched-pairs keeps the stereo image from deteriorating.
If a mastering engineer has a favorite analog EQ, or processor he wishes to use to create a particular sound, he should carefully balance out the cure versus the disease. There is always a loss in transparency when passing through analog stages, particularly A/D/A. Anyone who has patched processors in their consoles is aware of these tradeoffs. In other words, you have to carefully weigh the veil and fogginess that results from adding in an analog stage and additional converters with the subjective improvement from the processing versus processing the source in the digital domain. There will be an inevitable slight (or serious) veiling or loss of transparency due to each conversion. However, perhaps the mastering engineer feels the music will benefit from the sonic characteristics of a vintage compressor or equalizer...maybe he's looking for a "pumpy" quality that can't be obtained with any of today's digital processors (many people complain that digital processing is too "clean"...certainly a subject for another essay). There are many vintage "sounds" and other effects that still can only be obtained with analog processors. And finally, some mastering engineers claim that analog processors sound better than digital processors. I'm not one of them; I won't make that blanket statement. But I agree that analog processing is the "bees knees" for many musical productions. For example, I transferred a client's digital file to 1/2" analog tape and then back to 24-bit digital. Why? Because it sounded better. The analog tape stage did just the right thing to the source. I also had to make the fine choices of tape type, flux level, speed and equalization to help attain the spacious, warm, yet transparent sound quality my client and I were looking for. Ultimately, we used (and preferred) the analog dub to the original digital source for 8 out of the 10 tunes! Even without going through the analog tape, I have always maintained that A/D and D/A conversion processes are the most lossy part of the chain. When we do go back to the analog domain, I use the highest-quality D/A converter with low-jitter clocking, carefully calibrated levels, short analog signal paths and quality cables, and when converting back to digital, an extremely high-quality A/D converter. Then, the slight losses in transparency may be offset by the improvement due to the unique analog processing.
Our choice of whether to use analog or digital processing depends on the nature of the source, the music, and the tools we have on hand. I have some digital tools now which are so remarkable when used properly that clients with excellent ears cannot believe that the processing was not analog!

Before Mastering: Mixing, Editing, and Tape or File Preparation

Of course, before you get to the mastering stage, there is the mixing stage, which may be followed by an editing or processing stage. Many of you have purchased one of those new digital mixers to "stay in the digital domain" from beginning to end; many of you may have purchased a DAW (editing workstation) to prepare your tapes or files. Before Mixing: Please read my story, More Bits Please, which tells you how to use digital consoles and DAWs which mix, to their best advantage. Before editing or preparing your tapes for mastering, please read my article Preparing Tapes and Files for Mastering. You'll be glad you did.
Thanks for reading!

More Bits Please!

I. How To Pick The Right Digital Audio Workstation
II. How To Keep Your Sound Pure During Recording And Editing
III. The Source Quality Rule
IV. Detecting Those Sonic Bugs: Using the bitscope and other techniques to protect your precious data
V. Digital Consoles: How to Make a Better Mix with a Digital Console; Analog vs Digital Mixing
VI. No Longer The Missing Link: Affordable 24-bit File Interchange

I. Picking the Right DAW

"Faster, better, cheaper; pick only one of the three."

This adage is truer than ever in the age of digital audio recording. Occasionally you can get two out of three, but never all three at once. As computer power has become cheaper, more companies call themselves "manufacturers of recording hardware." It's now possible for a couple of guys to "invent" a Digital Audio Workstation in their basement out of a computer, an audio board, some mail order hard disks, and a little software glue. There are many startup companies trying to sell you the latest DAW-mousetrap, and with some flashy advertising, the world may beat a path to their door. Is the analog tape recorder dead? Have the days of precision-engineered mechanical parts and quiet roller bearings bitten the dust? Can you really get the quality of a $20,000 high-speed, widetrack analog tape recorder with the newest digital wonder consisting of computer, board and hard disk and costing less than $4000? How much should that quality really cost, with current computer technology? In this article, we'll try to separate your expectations from reality.

This article will take a look at DAWs, digital tape recorders and digital mixers in a fashion you may have never considered. First, the DAW...yes, it may slice and dice, but does it sound good? Before you buy the latest cheap box, don't forget that it takes a lot of talent and man-hours to produce good DSP software. One man-year is not enough time to produce a set of good sounding equalizers, a software digital mixer, mature editing tools, and recording and overdubbing tools. In five man-years, a talented set of individuals can create a working, reasonably dependable software-based system, and in ten man-years, a very sophisticated system. The key word is talented. The company producing this gear must have the right combination of skilled DSP engineers, user interface engineers, alpha-test supervisors, beta test supervisors, and a sufficient beta tester user base to give feedback. Because every computer program has bugs, lots of them. The trick is to turn them into little bugs before the program makes it to the street, where those bugs'll bite you. For we're not creating word-processing documents here, we're trying to make high-fidelity music. One misplaced bug in DSP code can produce subtle, or severe sonic fatalities.

5 x 1 Does Not Equal Five

So, the first rule in choosing a DAW is to be skeptical over the newcomers. Be wary of the one-year old company producing DAWs. In order for a one-year-old company to have the requisite five man-years of software development, they would need at least five very talented and coordinated DSP engineers. Coordinated, because during program development five people can easily get in each other's way; this can cause far more bugs (and missing features) than one software engineer working by himself for five years. In the case of software development, five times one does not always equal 5. So the one-year-old (or two-year old, or five-year old) company better be well-managed, with software engineers lured (or stolen) from their nearest competitors, excellent business capital (to survive those lean years and still be around to support the product you invested in), and lots of talent. But talent does not guarantee good product. Company management must be quality-oriented. When a large corporation wanted to get into the DAW market, very fast, they hired a crew of talented DSP engineers, but management cut corners in software development, in order to bring out the product in a year or so, and make dollars fast. Needless to say, that company's DAW division has made a rough start.

Learn everything you can about the company whose products you are about to invest in. A company which has been around five years and has a strong presence in the marketplace has a good potential of surviving. But maybe five years is not enough. A while back, a certain DAW manufacturer that had been around for five years was bought out by a large conglomerate, which soon decided to get out of the DAW market. Overnight, thousands of loyal users became owners of a white elephant. That's why I like 10-year-old companies even better....

Besides the obvious questions about development capital and financial stability, here are some other important technical questions you should ask before buying. Talk to the users (all ten of them?). How satisfied are they with the product, its performance, its potential, and most of all, its sound? Be very wise-don't rely on the company's "feature-promises". Don't expect the new ones to arrive as fast as the company predicts. All software manufacturers miss their deadlines and leave announced features out of their products. If leaping to conclusions were an Olympic event, software marketing directors would get gold medals every time. So if the product does not have the features you want today, don't buy it on the basis of "real soon now".

What Does It Really Cost?

Quality, features and reliability do not come cheap. The "BuzzSaw 2000" workstation you're considering may have reduced sound quality, features, and reliability. Man-hours of R&D really do cost. More realistically, instead of "a few thousand dollars", a robust workstation may require an investment from $8,000 to $20,000 especially if you want sophisticated video-synchronization features or high-quality noise reduction. Some manufacturers permit purchasing a system in incremental modules, so you may be able to get in on the ground floor of a quality system for less money.

It's Showtime!

Yes, check out the DAW's editing features. Make sure you can cut, paste, drag, drop, scrub, mix, and equalize. Talk to a user who's doing the exact work you are doing. A workstation that does well at video post may not be good with CD mastering. An editing station that's good for 60 second radio commercials may not be able to do long radio dramas. Watch over the user's shoulder. Get a real-world demonstration, not showroom hype. Are they demonstrating the release product, or a beta? How's the learning curve? Is it long or short? High power is often accompanied by a long learning curve, so you have to decide which is more important to you. Personally, I choose high power, even if the learning curve is longer, because the rewards are greater in the long run. But you may have lots of users at your company, and they all have to take a turn at the workstation. In that case, pick a DAW with a short learning curve.

A Sound decision...

It's a good start if the users give a DAW high marks for sonic quality. But ultimately the equipment has to pass the test of your ears. Shortly, I'll tell you how to perform an easy, foolproof listening test for sound bugs that you can perform on almost any DAW. Digital is digital, right? What goes in is what comes out, right? Not necessarily. My article The Secrets of Dither, describes how mixing, equalization, gain changing, and digital processing increase the wordlength of digital audio words. Your DAW has to be able to handle these operations transparently in order not to alter sound. The first requirement for good sound is 24-bit data storage and even higher resolution processing. If you want your music to lose stereo separation, depth and dimension, become colder, harder, edgier, dryer, and fuzzier, then don't look "under the hood".

II. Question Authority, or Perils Of The Digits

Let's see how you can keep the sound of your tape (or digital file) intact on its way to the CD Mastering House.

Let's discuss some digital do's and don'ts.

Mixing comes before editing.

So, before you edit, and before you mix, please read my article The Perils of Compression. After mixing, it's time to prepare your materials, and possibly edit.

If you mix to analog tape, it's best to make a safety digital copy, edit the analog (if necessary) with a razor blade, and send the original tape to the mastering house. A 30 IPS, 1/2" two-track tape contains a wide frequency and dynamic range, and is a superior recording medium. Some will argue that analog tape is more pleasant sounding than a digital recording (is that why are so many of us are nostalgic for the sounds of the 50's and 60's?). My essay called Back To Analog talks about those sonic differences. But the newer digital formats record at 24-bit, at sample rates up to 96 Khz and beyond (though the benefits of 192 kHz are controversial considering the human ear can hear nominally to 20 kHz). We are living in very interesting (and expensive) times. My Back To Analog essay makes some comments about the sound of 96 Khz/24 bit digital audio, and the article Audio Mastering refers to some dos and don'ts about digital versus analog processing.

So, when mixing, with few exceptions, we recommend that you keep your sound in the digital domain once it has crossed over the line. At the mastering house, using superior monitoring and experience, we can supply "just enough" warming or "sweetening" or whatever your mix may need to take it to that "finished quality". Experienced mix engineers know what their monitors and equipment are telling them and may choose to add some processes after mixing, but we recommend that you send both versions.

What about digital copying? Digital copying is ok. But what about digital editing, level changing, equalization or other processing in the digital domain? We recommend that you avoid going down multiple dsp generations, especially to add processes which are better left to the mastering stage. Please leave post-processes such as these to the mastering house. Here are some of the reasons why...

Question Authority...

No processor (analog or digital) is totally transparent. Try to keep your fragile mix from going downhill before sending it to mastering by avoiding additional DSP generations. Those little bits can undergo a perilous journey through some of the digital processors and editors on the market. If there's a DSP inside, suspect the worst until you know for sure. There are some tests you can perform on your digital processors and editors (or workstations) without expensive test equipment. These tests include linearity, resolution, and quantization distortion, common problems caused too-often by digital audio editors.

In other words, while you may be tempted to save time or money by doing preliminary editing with a digital audio editor, be very careful. A digital editor, after all, is just one big computer program; computer programs have bugs (there's not one bug-free program in existence!) and one of those bugs could be guilty of distorting your digits, in a big, or very subtle way. The sophisticated digital mastering systems at CD mastering houses also have bugs, but undergo regular testing to verify proper sound quality. We have received recordings with truncated fades (where the audio sounds like it dropped off a cliff!), distorted audio on the fadeouts; music with poor low-level resolution that is a shadow of its former self; music whose soundstage (stereo width and depth) appears to have collapsed, or recordings that have an indescribable "veil" over the sound compared with their sources. Here are some pointers that will help you avoid these problems:
Don't wreck your digital mix...

Always make a safety copy. Never send your only copy in the mail.

If you would like to try some post-mix processes on your already-mixed file (for example, eq, compression, tape emulation) be sure to send both versions to the mastering house. Perhaps we have a better or superior-sounding method of getting where you want to go. It's easy to fall in love initially with a squashed mix that later proves to be fatiguing and boring.

Please do not use any extra loudness processors in your mixing. Do not try to "compete" in level with any mastered product as you will actually defeat your purpose! It is very hard to make a squashed and overcompressed mix into a loud master. It's a lot easier to make a loud master from a clean, dynamic mix. Do not worry if you think your mix is not as "hot" as a current release. If your mix sounds good when you turn up your monitor, this is all you need to do. Most of the loudness-making tools take the sound downhill, require extreme skill to avoid degradation or distortion. At the mastering house we seek the most pristine, original source possible. No one should ruin your mix, especially you!

Always check the files you intend to send for mastering. If you made them via bounce (bounce to disk, aka "capture"), test your files by bringing the captures back into your editor and make sure you didn't upcut a beginning or miss an end. Good advice is to add 5 seconds to heads and tails... better safe than sorry. A five minute check on your part can save hours of grief later on.

Send the Unedited Original: Editing is like whittling soap. You can remove a piece, but you can't restore what's been chopped off! Yes, it's a good idea to make safety copies and put together some tests to find a good song order, even test fade-ins and fade-outs, but it's better to send the "raw" original, unfaded material to the mastering house (along with a good written log of where to find the cuts). Leave all the decay you can; there is less chance of degradation or missing a piece, and the mastering house probably has precise digital tools for performing artistic fades, or we might turn a fade into a segue (crossfade) if you like the idea. Also, fades which are performed in front of compressors can sound very different than fades performed after compression! If you have ideas on how the fades should be performed, give some suggestions to the mastering engineer or provide examples or alternate mixes with fades. Plus, there are things we can do that you may not have considered. For example, I've got some tricks that can create real-sounding endings on tunes that everyone thought had to be faded. There's even a bonus in sending the original raw mixes, as we now have available outtakes, alternate mixes (vocal up, vocal down, etc.) or other sections the mastering engineer can use to repair noises or problems you may not have noticed. The mastering engineer will order the tunes, carefully smooth fade-ins or fade-outs, place black or roomtone between the tunes, in extremely efficient time. Plus, at the mastering studio, each fade-out or level will be controlled with dither, a topic worthy of discussion.

Levels: Peak levels ideally should not exceed -3 dBFS on your meters. Sure you could go higher, but standard digital meters do not reflect intersample peaks which can be OVER 0 dBFS even if not shown on your meter. At 24-bit, you do not have to worry about signal to noise ratio and you will get a better result with a lower level and leaving some headroom for the mastering house. You would have to drop a 24-bit recording by 48 dB to reduce it to 16-bit resolution, so there's a lot of room---use it!

Noises: Alert the mastering engineer to any noises that bother you (note the time from first downbeat or from the beginning of your file). And we may be able to remove them with our noise-reduction processors, which include Cedar denoise and Cedar Retouch, Algorithmix, and TC Backdrop. And if the musicians talked before the ringout was over, or the bass player dropped his bow (shit happens), or the assistant stopped the recording before he was told, we can apply some of those techniques I mentioned to add convincing tails to a song that are indistinguishable from real life, and sometimes even better! It's a judgment call which noises are better left to be repaired in the mastering. If you can repair a noise by muting or fading down the instrument that makes the noise during the mixdown without creating an artifact, it's better for you to remove the noise. For example, noises made by a vocalist during a decay, where you can fade down or mute the vocalist's mike. Conversely, some noises might sound good if left in, producing a "relaxed, easy going feel" to an album. This includes countins, sticks, verbal comments by the musicians, and so on. Tell us the noises you like to keep, and we may find other noises that help to glue the album together.

If you would like to perform some complex editing prior to sending the material, TEST YOUR EDITOR first, also test it with a bitscope. Do this for each software revision. You really can't trust a manufacturer when your precious music is at stake. Listen carefully for degradation of soundstage width and depth, graininess, increased brightness or hardness. Listen on the finest reproduction system possible, or these changes may be perceived as too subtle and you won't know you've ruined your material until it's too late! You're welcome to send us a preliminary mix before you mix all your tunes. We will check it for tonal balance and for digital errors before you proceed.

Keep the bits. Cumulative digital processes if improperly performed can be very degrading to sound. The reason (and many engineers are not aware) is that almost every DSP computation adds additional bits to the wordlength. The wordlength can increase to 24-, 56-, or even 72-bits. The right thing to do is keep your newly "lengthened" words as long as possible, until the final stage, where we will dither them down to 16-bits for the CD. 16-bit dither should be reserved as a one-time only process at the end of the chain.
What makes the CD mastering house different?

All the processors at the CD mastering house produce 24-bit output words whenever possible. If the mastering engineer employs digital processing on your tape, he/she will endeavor to keep your tape in the 24-bit domain until the final stage. When properly applied, high resolution processes maintain a degree of warmth and space that is hard to believe. And that's why it can sound so good! A good, experienced mastering house tests each processor they use for resolution, distortion, jitter, and overall sound quality, auditioning in a superb acoustic with excellent monitor loudspeakers. Use the Mastering House like a mothership, ask us any questions you like, because our sole job is to make your recording the very best it can sound.

III. The Source-Quality Rule

This article is about getting "more bits" into our recordings, but there's a powerful opposite pressure to use an inferior-sounding, low-bit-rate (data compressed) delivery medium for home audio, radio, and for the Internet. Personally, I wish lossy data compression could be outlawed; while that won't happen, at least let's keep on lobbying for sound quality. One way to maintain quality is to follow this important rule: Source recordings and masters should have higher resolution than the eventual release medium. There's always a loss down the line, due to cumulative processing and lossy transmission techniques. For example, consider a lossy medium like the analog cassette. Dub to cassette from a high quality source, like a CD, and it sounds much better than a copy from an inferior source, like the FM radio. In other words, the higher the audio quality you begin with, the better the final product, whether it's an audiophile CD, a multimedia CD-ROM, or a talking Barbie doll.

Get ready for high-resolution release media (DVD, Blu-Ray, etc.) by following this source-quality rule. Prepare your masters now with longer wordlength storage and processing, and if possible, high sample rates. The 96 kHz/24 bit medium has even more analog-like qualities, greater warmth, depth, transparency, and apparent sonic ease than 44.1 kHz. Perhaps it's due to the relaxed filtering requirements, perhaps it's due to the increased bandwidth-regardless, the proof is in the listening. Therefore, produce your master at the highest resolution, and at the end (the production master), use a single process to reduce the wordlength or sample rate. Multiple processes deteriorate quality more than a single reduction at the end. The result: better-sounding Masters.

Another advance in the audio art is double-sampling processing. The improvement is measurable and quite audible, more...well... analog. Double (and higher) sampling sounds better when applied to compression and possibly with digital equalization. Dr. James A. (Andy) Moorer of Sonic Solutions, writes "[in general], keeping the sound at a high sampling rate, from recording to the final stage will...produce a better product, since the effect of the quantization will be less at each stage". In other words, errors are spread over a much wider bandwidth, therefore we notice less distortion in the 20-20K band. Sources of such distortion include cumulative coefficient inaccuracies in filter (eq), and level calculations.

88.2 kHz Reissues Will Sound Better Than The CD Originals

The above evidence implies that record companies are sitting on a new goldmine. Even old, 16-bit/44.1 session tapes can exhibit more life and purity of tone if properly reprocessed and reissued on a 24-bit/ 88.2 kHz (or 96 kHz) DVD. In addition, by retaining the output wordlength at 24 bits, it will be unnecessary to add additional degrading 16-bit dither to the product. Many of these older 16-bit tapes were produced with 20-bit accurate A/Ds and dithered to 16 bits; they already have considerable resolution below the 16th bit.

DSD versus Linear PCM

Sony's high-resolution DSD format is a one-bit (Delta modulation) system running at 3 Mbyte/second. The jury is still out on whether this system sounds as good as or better than linear PCM at 96 kHz/24 bit, but regardless, Sony's whole purpose was to follow the source quality rule. The company feels that DSD is the first medium that will preserve the quality of their historic analog sources, and that DSD is easily convertible to any "lower" form of linear PCM. Regardless of whether DSD or linear 96/24 becomes the next standard, it's a win-win situation for fans of high-resolution recording.

PCM:

Short for pulse code modulation, a sampling technique for digitizing analog signals, especially audio signals. PCM samples the signal 8000 times a second; each sample is represented by 8 bits for a total of 64 Kbps. There are two standards for coding the sample level. The Mu-Law standard is used in North America and Japan while the A-Law standard is use in most other countries.
PCM is used with T-1 and T-3 carrier systems. These carrier systems combine the PCM signals from many lines and transmit them over a single cable or other medium.

PCM is also the usual digital method used for music audio playback of music CDs. While supported by DVDs, DVDs have a greater volume so they use Linear PCM, which has a higher sampling rate — up to 24-bit at a sampling rate of 96 kHz.

Also:

LPCM is a particular method of pulse code modulation which represents an audio waveform as a sequence of amplitude values recorded at a sequence of times.

LPCM specifies that the values stored are proportional to the amplitudes, rather than representing say the logarithm of the amplitude, or being related in some other manner. In practice these values will be quantized. Theoretically, there is no loss or error in conversion and reconstruction, as long as the sampling rate is just over twice the highest desired frequency component of the recorded signal [Claude Shannon; Harry Nyquist]. For example, if you want to record audio at up to 20 kHz, you would need a frequence of sampling (F/s) of a little more than 40 kHz. Some argue that 60 kHz F/s would be better as the rolloff frequency of the low pass filter which attenuates the echoes of the recorded signal, which otherwise would beat against the original, at multiples of the sampling rate, would be ultrasonic and therefore not smear the high end of the converted (audible range of the) audio. A stable clock, for conducting the regular timing of the sync pulses, which are scaled to represent the amplitude values of the original analog signal at each sampling instance, is essential to the good functioning of the digital audio system.

Implementations

LPCM is the method of encoding generally used in conjunction with the WAV container format, the de facto standard for uncompressed audio on PCs. The term PCM and LPCM often refer explicitly to the format used in WAV files, though LPCM data may also be commonly stored in other formats such as AIFF. LPCM is further used for the lossless encoding of audio data in the compact disc Red Book standard. LPCM has been defined as a part of the DVD standard. AES3 is a particular format using LPCM.

Standard sampling resolutions and rates

Common sample resolutions for LPCM are 8, 16, 20, 24 or 32 bits per sample.
While two channels (stereo) is the most common format, some can support up to 8 audio channels (7.1 surround).
Common sampling frequencies are 48 kHz as used with DVD format videos, or 44.1 kHz as used in compact discs. Sampling frequencies of 96 kHz or 192 kHz can be used on some newer equipment, with the higher value equating to 6.144 megabits per second per audio channel.

DVD standards

Most DVD players only support 48 kHz/16-bit capability. Only more high-end players have built-in 96 kHz/24-bit capabilities. The DVD-Audio standard supports 192 kHz/24-bit playback.

Signal Chains

One obstacle to better sound is our need to chain external processors and perform capturing and further processing in our workstations. Even if manufacturers use internal double precision (48-bit) or triple precision (72-bit) arithmetic, the chain of processors must still communicate at only 24 bits, for that is the limit of the AES/EBU standard. Despite that, I welcome manufacturers who use higher precision in their internal chains, because all other things being equal, we'll have better sound. The ultimate solution would be to extend the AES/EBU transmission standard to a longer wordlength, but with care we can still get excellent results by using longer internal wordlengths and truncating (or preferably dithering) down to 24 whenever we have to. When using plugins within a native structure, the wordlength is retained at 32 bits float (or 64 bit in some machines) which also reduces cumulative degradation.

Floating or Fixed?

Don't get into a misinformed "bit war" confusing floating point specs with their fixed point equivalent. A 32-bit floating point processor is roughly equivalent to a 24-bit fixed point processor, though there are some advantages to floating point. With the 40-bit floating point processors, and all things being equal, they seem to sound better than the 32-bit versions (but when was the last time all things were equal?). On the fixed point side, the buzz word is double-precision, which extends the precision to 48 (fixed point) bits. Double precision arithmetic (or doubled sample rate) in a mixer requires more silicon and more software to have the same apparent power, that is, the same quantity of filters and mixing channels. It'll be expensive, but ultimately less expensive than its high-end analog equivalent, a mixer with very high voltage power rails, and extraordinary headroom (tubes, anyone?).

Warm or Cold? Digital is Perfect?

What does a double-precision digital mixer sound like? It sounds more like analog. The longer the processing wordlength, the warmer the sound; music sounds more natural, with a wider soundstage and depth. Unlike analog tape recording and some analog processors, digital processing doesn't add warmth to sound, longer wordlength processing just reduces the "creep of coldness". The sound slowly but surely will get colder. Cold sound comes from cumulative quantization distortion, which produces nasty inharmonic distortion.

That's why "No generation loss with digital" is a myth. Little by little, bit by precious bit, your sound suffers with every dsp operation. As mastering engineers who use digital processors, we have to choose the lesser of two evils at every turn. Sometimes the result of the processing is not as good as leaving the sound alone.

IV. Detecting Those Sonic Bugs

Did you know that the S/PDIF output of the older Yamaha mixing consoles is truncated to 20 bits? Now how did I know that? Because I tested it! And you can, too, with some very simple equipment. There are some legitimate reasons why Yamaha made that choice, although I do not agree with them. This means that if you want to get all 24 bits out of your Yamaha console, you must use the AES/EBU output. There are simple ways to adapt the Yamaha's AES/EBU output to the S/PDIF input of your soundcard, and this will preserve all the bits. Many (if not all) soundcards that work at 24 bits accept the 24 bits on their S/PDIF inputs.

Proper use of those 24-bit words is equally important. Bugs that affect sound creep into almost every manufacturer's release product. In 1989, the latest software release of one DAW manufacturer (whose machine I no longer use) had just hit the market. I edited some classical music on this workstation. There was a subtle sonic difference between the source and the output, a degradation that we perceived as a sonic veil. Eventually it was traced to a one bit-level shift at the zero point (crossover point, the lowest level of the waveform) on positive-going waves only. This embarrassing bug should have been caught by the testing department before the software left the company. Does your DAW manufacturer have a quality-control department for sound, with a digital-domain analyzer such as the Audio Precision? Do they test their DSP code from all angles? Incredible diligence is required to test for bugs. For example, a bug can slip into equalizer code that does not affect sound unless the particular equalizer is engaged. It's impossible to test all permutations and switches in a program before it's released, but the manufacturer should check the major ones.

A Bitscope You Can Build Yourself

The first defense against bugs is eternal vigilance. Listening carefully is hard to do-continuous listening is fatiguing, and it's not foolproof. That's why visual aids are a great help, even for the most golden of ears. In the old days, the phase meter was a ubiquitous visual aid (and should still be a required component in every studio); our studio also uses a product we call the "digital bitscope", that is easy and inexpensive to put together. It's not a substitute for a $20,000 digital audio analyzer, but it can't be beat for day-to-day checking on your digital patching, and it instantly verifies the activity of your digital audio equipment. Think of it this way: The bitscope will tell you for sure if something is going wrong, but it cannot prove that something is working right. You need more powerful tools, such as FFT analysers, to confirm that something is working right.

However, the bitscope is your first line of defense. It should be on line in your digital studio at all times. You can assemble a bitscope yourself--see The Digital Detective. If you're not a do-it-yourselfer, Digital Domain manufactures a low-cost box that can be converted to a bitscope with the addition of a pair of outputs and a 2-channel oscilloscope. Our bitscope is always on-line in the mastering studio. It tells us what our dithering processors are putting out, it reveals whether those 20-bit A/D converters are putting out 20-bit words, and it exposes faults in patching and digital audio equipment.

Some Simple Sound Tests You Can Perform on a DAW

With the output of my workstation patched to the bitscope, I can watch a 16 or 20-bit source expand to 24-bits when the gain changes, during crossfades, or if any equalizer is changed from the 0 dB position. A neutral console path is a good indication of data integrity in the DAW. After the bitscope, your next defense is to perform some basic tests, for linearity, and for perfect clones (perfect digital copies). Any workstation that cannot make a perfect clone should be junked. You can perform two important tests just using your ears. The first test is the fade-to-noise test, described previously in my Dither article.

The next test is easier and almost foolproof-the null test, also known as the perfect clone test: Any workstation that can mix should be able to combine two files and invert polarity (phase). A successful null test proves that the digital input section, output section, and processing section of your workstation are neutral to sound. Start with a piece of music in a file on your hard disk. Feed the music out of the system and back in and re-record while you are playing back. (If the DAW cannot simultaneously record while playing back, it's probably not worth buying anyway). Bring the new "captured" sound into an EDL (edit decision list, or playlist), and line it up with the original sound, down to absolute sample accuracy. Then reverse the polarity of one of the two files, play and mix them together at unity gain. You should hear absolutely no sound. If you do hear sound, then your workstation is not able to produce perfect clones. The null test is almost 100% foolproof; a mad scientist might create a system with a perfectly complementary linear distortion on its input and output and which nulls the two distortions out but the truth will out before too long.

If the workstation is 24-bit capable, and your D/A converter is not, you may not hear the result of an imperfect null in the lower 8 bits. Use the bitscope to examine the null; it will reveal spurious or continuous activity in all the bits and tell you if something funny is happening in the DAW. Even if your DAC is 16 bits, you can hear the activity in the lower 8 bits by placing a redithering processor in front of your DAC.

Use the powerful null test to see whether your digital processors are truly bypassed even if they say "bypass". Several well-known digital processors produce extra bit activity even when they say "bypass"; this activity can also be seen on the bitscope. Use the null test to see if your digital console produces a perfect clone when set to unity gain and with all processors out (you'll be surprised at the result). Use the null test on your console's equalizers; prove they are out of the circuit when set to 0 dB gain. Use the null test to examine the quantization distortion produced by your DAW when you drop gain .1 dB, capture, and then raise the gain .1 dB. The new file, while theoretically at unity gain, is not a clone of the original file. Use the null test to see if your DAW can produce true 24-bit clones. You can "manufacture" a legitimate 24-bit file for your test, even if you do not have a 24-bit A/D. Just start with a 16-bit or 20-bit source file, drop the gain a tiny amount and capture the result to a 24-bit file. All 24 of the new bits will be significant, the product of a gain multiplication that is chopped off at the 24th bit. You'll see the new lower bit activity on the Bitscope.

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Inject Protools Data:

TECHNICAL WHITE PAPER

This technical article provides detailed information about how the Pro Tools mixer operates. In so doing, we will demonstrate its summing characteristics and explain how a 48-bit “clean” mixer functions within the 24-bit TDM (Time Division Multiplexing) environment. By providing some ‘behind-the-scenes’ information about mixing and summing in Pro Tools, we hope to shed light on a few myths about mixing ‘in the box’ with Pro Tools, as well as provide you with a better understanding of the mechanics of summing signals.

Though the tasks of any digital mixing system are the same—combining audio streams without clipping while also retaining their low level detail—the approaches used can be very different. This article provides a description of the approach taken in Pro Tools and the benefits of using this system.

WHERE ARE THE BITS AND HOW ARE THEY USED?

Let’s first define some terms that we’ll be using in this discussion.

• Bit: A binary digit. In digital audio, the word length of the system indicates how many bits are applied to recording the sound and used for calculating changes such as level, EQ, dynamics, etc. A single bit represents the smallest change in the signal. Large digital words provide more discrete values so the changes represented in the smallest or least significant bit (LSB) can be very slight. Bits are also used to describe the dynamic range of the system. A single bit represents about 6.02 dB of dynamic range.

• Resolution: The number of discrete values available in a digital system. In figure 1, you can see the number of discrete steps each number of bits yields. More discrete steps allow for very fine changes in the signal to be faithfully reproduced.

• Decibel: A logarithmic scale for representing the ratio of two amounts of power. Here, we’re using the decibel as the unit of measurement for dynamic range—the difference between the highest power signal and the smallest power signal of a recording.

Now, let’s take a look at how Pro Tools takes signals from the analog world, converts them to digital signals, and follow them through the system to where they emerge again as reconstructed analog audio.

Check the rest here: http://akmedia.digidesign.com/support/docs/48_Bit_Mixer_26688.pdf

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V. Digital Consoles - How to make a better mix with a Digital Console; Analog vs digital mixing

Let's discuss the use of digital consoles with digital recorders. Knowing how to use this gear really separates the men from the boys. Digital consoles suffer from the same wordlength and truncation problems as DAWs. Truncation without redithering is always bad, but depending on where you truncate, the result can be sonically benign, or very nasty. For example, truncating a 24-bit A/D to 16 bits is relatively benign because most mike preamps are noisy enough to provide some dithering action. But using a DSP to drop gain only .1 dB in a console and then truncating the output to 16 bits is very damaging, shrinking soundstage and producing harsher sound. Be aware of these facts when using digital consoles with digital recorders. Always use dither to reduce the console's long wordlength to the recorder's wordlength. If your digital console does not have dithering options, you'll be better off with a very high-end analog console. That's one of the things that separates the higher priced digital consoles from the cheap ones. Cheap digital consoles do cost--you pay in reduced sound quality.

There's an engineer on the leading edge, who had been working with 24-bit recording and a digital console, but reverted to a purist-quality analog console when he upgraded his converters to 24 bits. He found he got better-sounding results mixing live sources in analog and then feeding the 24-bit A/D than by starting with A/D's and feeding a digital console. It takes a very special digital console to preserve 24-bit quality; it's also difficult and expensive to design an A/D converter that retains high resolution inside the polluting environment of a digital console.

VI. No Longer The Missing Link -

Affordable 24-bit file interchange

We encourage clients to send us 24-bit files at the highest sample rate you are working at. See my article Preparing Tapes and Files for Mastering for descriptions of all the new high-bit formats.

We Love Digital and Analog.

We also Love Hardware and Software.

VII. Conclusion

DAWs, digital tape recorders and digital consoles affect sound. Use these tools properly and your music will sound better. Mastering houses thrive on high-resolution sources. Consider the choices and send the best source you can for mastering. Manufacturers--Give Us More Bits--and please, make them compatible!

Our very good friend and Aerospace Tech Head - Paul Fargo - The "Manley Answer Man"...

ALTIVERB has to be some of the best sounding Reverb on the Planet. Just kick Royally!

Lexicon and Eventide. There just is no substitute..

True Systems Just "Wow" Clean and Beautiful.

This is the Catz Meow

And this BCL is my real secret weapon. I use this at all times and in endless ways...

~ Click on the Picture above for Information on "Separations" Mastering Techniques ~