It's not how loud you make it, it's how you make it loud!

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.

{slide=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.
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{slide=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.
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{slide=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.
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{slide=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.
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{slide=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!
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{slide=Unique Digital Processes}

There are also some unique techniques that I cam perform only in the digital domain, one of which I call microdynamic enhancement , another is tonalization and the third is my patented K-Stereo Process (which is available to other mastering engineers).

For example, microdynamic enhancement can restore or simulate the liveliness and life of a great live recording. I've used it to get more of a big-band feel on a midi-sample-dominated jazz recording. I've used it to put life into an overly-compressed (or poorly-compressed) rock recording. It's really useful and extraordinary in helping to remove the veils introduced in multi-generation mixdowns, tape saturation and sound "shrinkage" that comes from using too many opamps in the signal path.

K-Stereo finds a use with unidimensional (flat-sounding) material, often the sad result of low-resolution recording and mixing. It is very different from the various width-altering processes that are now-available. K-Stereo captures and brings out the original ambience and definition in a source. The degree of stereoization is completely controllable. Instruments in the soundfield obtain a more natural space around them. The process is totally natural, utilizing psychoacoustic principles, and it's fully mono-compatible. For more information on this remarkable process, check out the K-Stereo.

DSP engineers are constantly inventing new ways to simulate all the traditional analog processes. So there's a lot to be said for digital processing, and I have no doubt that will become the dominant audio mastering method over time. For the forseeable future, in many cases we use a hybrid of analog and digital processing techniques to produce the best-sounding master.
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{slide=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.
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Thanks for reading!

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The Pros and Cons of Recording and mixing via Analog versus Digital

This article has been revised and updated from an editorial counterpoint which appeared in Pro Sound News, January 1997. Here's a refreshing alternative perspective to what's going on in the studio scene for everyone from musicians to owners of project studios to large studios.

Analog vs. Digital - The Good, the Bad, and the Ugly
Doing analog audio in the sixties and seventies was hell. Most of us would like to throw our bias oscillators in the garbage. Analog requires constant vigilance to sound good. In addition, you can't copy an analog tape. The second generation just falls apart; it's a pale replica of the first. If analog's so bad, what's the problem with digital recordings? Can't we give them the warmth of analog if we use vintage tube mikes and analog processors? There must be something to that argument, or the whole industry wouldn't be doing the retro-tube trip. But I wonder if we're all doing it for the wrong reasons. Please remember that there's good tube equipment out there, and a lot of bad. There's also good digital equipment and an awful lot of bad. Much tube equipment is overly warm, fuzzy, noisy, unclear and undefined. Only the best-designed tube equipment has quiet, clear sound, tight (defined bass), is transparent and dimensional, yet still warm without being artificial or muddy. Similarly, some of the cheap digital audio equipment is edgy or hard-sounding, dimensionless, and unclear. Only the very best digital audio equipment (and it's getting better every day) can lay claim to good soundstage width and depth, purity of tone without an artificial edge, and transparency.

Bad Digital versus Good Digital
Many people have argued that digital audio recording is more accurate than analog, saying the accuracy of digital is why we're noticing hardness and edginess in our recordings, and have regressed to tube and vintage microphones. That's only a half-truth. Let's distinguish between bad digital and good digital equipment design. Bad digital sounds bad because it is bad. Bad digital equipment has distortions that innately increase edginess and hardness. Edgy sound can be caused by many factors: sharp filters, poor conversion technology, low resolution (short wordlength), poor analog stages, jitter, improper dither, clock leakage in analog stages due to bad circuit board design and many others. Placing sensitive A/D and D/A converters inside the same chassis with motors and spinning heads is also a dangerous practice. It takes a superior power supply and shielding design to make an integrated digital tape recorder that sounds good.

A lot has changed since I first wrote this article in 1997. Coincident with the demise of the tape-based multitrack and 2-track have come better converters and higher sample rates. "Bad Digital" has largely gone away and been replaced with ranges from "just acceptable" to "very good". There was a time when I could attribute the edgy-sounding, dimensionless mixes that I received to the poor converters in the MTRs (multitrack recorders), but now we see losses in a matter of degree. Nowadays many MTRs record at 48 kHz and above and wordlengths of 24 bits. The DAWs record at 24-bit fixed, and mix at 32 to 64-bit float. The sonic performance of the lower-cost MTRs may not equal that of high-class separate converters, but by 2005 or so I could finally say that even an MTR in the $2000 range using internal converters performs pretty well. Everything has (thankfully) come up several notches over 1997-performance and down in price! Now when I get dimensionless, distorted or gritty mixes I usually attribute them to operator abuse of digital technology, overuse of low-resolution plugins and processing and overuse of cheap samples or synthetic instruments, for example, digital drumsets instead of the real thing. The A/D converter still contributes a bit to the sound quality, let's say 10%, even less if you work at 96 kHz, where even a mediocre converter sounds better.

Through loving care and a number of proprietary processes in the mastering stage, I can bring these mixes up to a much better quality level. It is possible to give the sound greater apparent transparency, more spaciousness, increased purity of tone, improved dynamics and transient response (where these changes are esthetically appropriate). A mastering engineer who has made and heard the best recordings can do a lot for weak mixes. But let's not forget the sound that can come from analog tapes mixed through analog consoles, and from widetrack analog masters. After reading this article, I think you'll reconsider the analog alternative.

Bandaids Instead of Cures
Bad digital benefits from the use of tube mikes and preamps because their warmth and noise help cover up the hardness of the rest of the signal chain. Use of warm-sounding mikes and preamps can become a fuzzy blanket that hides the potential resolution of the system, but it is not a cure, it is a bandaid. Even good digital benefits from proper choice of microphones and preamps (including well-designed tube equipment). Digital recording is considered to be "accurate," but each of its specs must be considered carefully. Consider its linear frequency response. With bad digital technology, linearity of frequency response can turn from virtue into a defect. We can no longer tolerate the distortion and brightness of some solid-state equipment (including poor A/D converters, microphones and audio consoles) because digital recording doesn't compress (mellow out) high frequencies as does low speed (15 IPS) analog tape. To summarize: digital recording can sound edgy for two reasons. One is linear frequency response, which reveals non-linearities in the rest of the chain. The other is built-in distortions in the A/D/A conversion process.

The Virtues of Analog Recording
Listening to a first generation 30 IPS 1/2" tape is like watching a fresh print of Star Trek at the Astor Plaza in New York. I believe that a finely-tuned 30 IPS 1/2" tape recorder is more accurate, better resolved, has better space, depth, purity of tone and transparency than many digital systems available today. Empirical observations have shown that you need a nominal "24-bit" A/D to capture the low-level resolution of 1/2" 30 IPS (if truth be told, the best converters only approach about 19-20 bit resolution in practice). It can also be argued that 1/2" tape has a greater bandwidth than 44.1 KHz or 48 KHz digital audio, requiring even higher sample rates to properly convert to digital. Listening tests corroborate this. 30 IPS analog tape has useable frequency response to beyond 30 KHz and a gentle (gradual) filter rolls off the frequency response. This translates to more open, transparent sound than any 44.1 kHz/16 bit digital recording I've heard. 1/2" 30 IPS analog tape has lots of information, like high resolution 35 mm film. 16-bit 44.1 KHz digital is like low-resolution video. As higher resolution (e.g. 96 Khz/24 bit) digital formats become the standard, maybe then we'll be able to say that digital recording is better than analog. But don't be fooled by the numbers; there's still some "magic" in the coloration of analog tape that we have not yet been able to reproduce in an all-digital recording, especially for popular music forms that often crave the sound of tape saturation. Analog tape has its own problems, but when operated within its linear range, unlike digital recording, it has never been accused of making sound "colder." However, digital recording has finally gotten good enough so that in acoustic music formats like classical and folk, some engineers are preferring digital recording's transparency over analog's warmth.

The Real Cure
If you want to improve the sound of digital recording beyond that of the built-in converters in the multitracks, you will have to spend a small fortune. You can add external A/Ds and D/As which may cost several times the price of the basic machine. To put it in perspective, The entire multitrack digital recorder costs less than a 2-channel A/D converter from the best audio firms! This points out the large economic disparity between "bad" and "good" digital. It's obvious that to have the best digital sound, your project studio can quickly become a high-dollar venture. This is analogous to the art of photography, where a single lens can cost 3 to 4 times that of the camera body.

At first glance it may seem that using a digital console to mix down from MTR can be an advantage, because you are not using the poor D/A converters in the MTR, but now you will have to deal with the long wordlengths produced by the calculations in the digital console, and the digital console has to get its internal extra-long wordlengths down to 24 bit with minimal distortion. Since distortion is cumulative, I suggest you minimize multiple passes through the DSP circuitry in the console or DAW. Numeric precision problems in digital consoles produce problems analogous to noise in analog consoles. However, there is a difference between the type of noise produced in analog consoles and the distortion produced by numeric problems in digital consoles. Noise in analog consoles gradually and gently obscures ambience and low-level material and usually does not add distortion at low levels. Numeric problems in digital consoles can cause several problems. Rounding errors in digital filters act much like analog noise, but at other critical points in the digital mixing process, wholesale wordlength truncations can cause considerable damage, destroying the body and purity of an entire mix, creating edgy sound, which audiophiles often call "digititis." Depending on the quality and internal precision of the digital console and digital processors you choose, and the number of passes through that circuitry, it might have been better to mix down to analog tape through a high-quality analog console.

If you do not use an analog mixing console in conjunction with "old fashioned" analog equalizers and processors--you'll have to take extra pains to keep your digital system from sounding cold. If you can't afford high-quality external A/Ds (and large hard drives), be careful of using too many tube bandaids. Tubes can cover up the evils of the cheap A/D/A's and processors, resulting in a warm, fuzzy sound, but that's preferable to a hard and edgy one. In other words, good digital is expensive and probably the best you can get from bad digital with bandaids is "warm and fuzzy!"

I still suggest trying analog tape! Invest in a great analog recorder. Your first step is to get a good two-track 1/2" machine. After that, consider using wide-track analog multitrack for the main tracks. To get good analog sound that's perhaps better than the most expensive digital, practice your alignment techniques, don't bounce tracks, use wider track widths and higher speeds than you did before.

Cost-wise, analog tape recording has finally exceeded that of even world-class AD/DA converters due to the extreme cost of tape. If you're not doing analog tape, try to get the very best converters and minimize mixdown passes through low-resolution processing and plugins.

Making the Right Trade off Decisions If you must choose some digital storage and processing, evaluate the tradeoffs carefully. Both analog and "good digital" are clear, detailed, warm, spacious, and transparent. We have to reevaluate the tradeoffs each year. For example, the cost of 2-track, 96/24 digital recorders has plummeted. I recommend the Tascam DVRA1000HD DVD recorder---with or without external converters it has raised the bar in 2-track machines. Study the compromises and look at each situation as a tradeoff: If you have too much "digital", and not enough "analog", your results will not be "fat" or "warm" enough. Or vice-versa! So, the well-balanced "hybrid" approach beckons, perhaps tracking with analog tape and mixing digitally, or vice-versa.

With today's choices, you can offer musicians a real value that sounds great. You can easily assemble an affordable multitrack system that sounds better than the old 44.1/16 systems. Hybrid can sound great! I'm looking forward to seeing your fabulous tapes or files at our mastering house!

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Copyright Digital Domain, Inc. We invite you to link to our site, which will be periodically revised.

{mov}Bass_Frequency_Surgery_mov{/mov}

 

This instructional video describes the fastest way to use an equalizer to fix a "one note bass" problem in a recording.
To download a higher resolution video with 16-bit, 48 kHz audio, please click here!

Part I - Technical Guidelines for the Uses of Compressors
We define compression, and give technical guidelines for its use in mixing and mastering.
Part II - The Perils of Compression
The esthetic ramifications of compression, the current loudness race, the bleak future of DVD audio quality, and more to the point, what to do about it!
Part III - Tools To Help Keep Us From Overcompressing
Including an honor roll of CDs that sound very good and aren't overcompressed.

A U.K. engineer who discovered our web site wrote.

I had a look at your web page and what a page! Exactly a page I have been looking for! Could you add something about using compression, like how much to do by yourself and how much to leave to the mastering lab?

First, let's define:
Com-pres'-sion 1) Reduction of audio dynamic range, so that the louder passages are made softer, or the softer passages are made louder, or both. Examples include the limiters used in broadcasting, or the compressor/limiters used in recording studios.
2) Digital Coding systems which employ data rate reduction, so that the bit rate (measured in kilobits per second) is less. Examples include the MPEG (MP3) or Dolby AC-3 (now called Dolby Digital) systems.

This article is about #1 above, dynamic range compression. To avoid confusion, please encourage people to stop using the word compression and use the proper terms "Data Reduction System" or "Coding system" for definition #2.

Now, for the basic two rules:

Rule #1: There are no rules. If you want to use a compressor/limiter of any type, shape and size in your music, then go ahead and use it.

Rule #2: When in doubt, don't use it!

How can you tell when you have enough compression?

• Discussing sound in print is like describing colors to a blind person, but let me try. Here's a simplistic example... supposing there are two sonic qualities of music, one called punchy, the other smooth... Let's say that some music sounds better punchy, other music sounds better smooth. Let's also assume for this example that you can achieve punchy or smooth sound through different amounts and types of compression, or not using compression at all.
• In general, try to avoid overall compression in the mix stage if:
  • you're mixing punchy music (the type of music that needs punch), perhaps using some individual compression on certain instruments or singers--and the mix already sounds punchy (good) to you.
  • you're mixing smooth music, and your mix already sounds smooth.
  • you play a well-recorded CD of similar music, and your CD in the making already sounds good (or better than) the CD in the player.
  • your music already seems to accomplish the sound you are looking for.

We'll discuss compression esthetics in more detail in Part II.

Technical reasons to avoid overall compression on your album
Save decisions on overall compression and individual tune equalization for an expert CD mastering house because:

1. The mastering house will have a more appropriate compressor with the proper attack, ratio, and release times exactly right for your music. If you mixed to digital tape, they will probably use a 24-bit digital compressor for the purpose.
2. They will likely be more experienced than you about the compromises, advantages and disadvantages of applying overall compression.
3. The mastering house can program that compressor with precision, adjusting it optimally for each tune in question. You're working out of context (without having the perspective of the entire album) by attempting these sorts of decisions during mixing.
4. The mastering house will be able to monitor your "CD in the making" using a calibrated monitoring system so that they know exactly how loud your "CD in the making" is compared to other CDs of similar music. For more information on loudness, see my article Level Practices in Digital Audio.
5. A good mastering house will be able to do all of this in a non-destructive, non-cumulative manner. In other words, after making a reference CD, they will be able to undo anything you are unhappy with, whether it be compression, EQ or levels. Whereas, most digital audio editing stations can only perform destructive EQ or compression, only with 16-bit word length, with a consequent loss of resolution as long internal words are either dithered (resulting in a veil if further processed), rounded (slightly better than truncated), or truncated to 16 bit. For further information, see my article The Secrets of Dither.
6. For the same technical reasons, it is not a good idea to use a digital compressor (or any digital processor) on your material before sending it for mastering. If you do feel the need to insert one of these boxes, for example, to give a demo CD to your client, be sure to also make a non-processed version to prepare for the mastering house. It is likely that the mastering house will have a fresher-sounding, more effective approach at polishing your material, and it's self-defeating if they have to try to undo what was done.
7. If you apply overall compression to your music, and your choice of compressor was wrong (e.g., the compressor you chose caused subtle pumping or breathing, loss of transients, loss of life or liveliness, etc. These are typical symptoms of "compressor misuse" on tapes I have received), the mastering house will have a difficult or impossible time attempting to undo the damage. As I've mentioned, mastering is like whittling soap; it is hard to undo compression. However, I do have some tricks up my sleeve (grin) that can restore some life to squashed tapes.

Part II
The Perils of Compression
or
The Ghost of CD Past

Introduction
24-bits, 96kHz, multichannel sound. These are some of the exciting features of the upcoming super audio disc. Before we can use these new capabilities to their fullest, we must learn not to repeat our past mistakes. Some of our engineering practices with the CD have done a serious disservice to the consumer. This article looks at one of those practices--overcompression of dynamics, and makes some recommendations that will turn the DVD (and new 16-bit CDs) into the true media of the future.

Dynamic Range - The Ups and Downs of Music
Before we can study the art of compression, we must learn to appreciate the power of music's dynamic range. How does music grab our interest? For short periods (about the length of a "single" played on the radio or in the disco), power and loudness can grab our initial attention. But at home, variety of dynamics maintains our interest for long periods of time. Good music written for a long-term musical experience contains a judicious mixture of variety and similarity in dynamics. A production which is relentlessly loud (or relentless in its sameness) can become boring very fast. At the age of 10, I learned the lesson of Franz Joseph Haydn's Surprise Symphony, the first composition to teach the importance of dynamic contrast. Musical genres that depend on constant sameness become old very fast. Disco died because it became boring, and I'm convinced that overcompression (which eliminates dynamic contrast) contributed to its death, by creating a continuously loud, boring dynamic. I wonder if the current slack in music sales is related to overcompression and its tendency to give everything a monotonous loudness--is the public voting against compression with its pocketbook?

Any genre that does not grow in musicality will quickly die, and dynamic contrast plays a big role in musicality. Today's Rap music has taken a 250-year-old lesson from classical composition, by beginning to incorporate a melodic and harmonic structure. The genre can further grow and avoid sounding tiresome by expanding its dynamic range, adding surprises. Silence and low level material creates suspense that makes the loud parts sound even more exciting. Five big firecrackers in a row just don't sound as exciting as four little cherry bombs followed by an M80. This is what we mean by dynamic range. Radio, TV and Internet distribution are currently too compressed to transmit the joy of wide dynamic range, but it sure turns people on at home, and also in the movie theater.

Films provide an ideal framework to study the creative use of dynamic range. The public is not consciously aware of the effect of sound, but it plays a role in a film's success. I think the movie The Fugitive succeeded because of its drama, but despite an aggressive, compressed, fatiguing sound mix. From the beginning bus ride, with its super-hot dialog and effects, all the crashes were constantly loud and overstated, completely destroying the impact of the big train crash. I can hear the director shouting, "more more more" to the mix engineers. Haven't they heard of the term "suspense?" In contrast, the sound mix of Titanic, is a masterpiece of natural dynamic range. The dialog and effects at the beginning of the movie are played at natural levels, truly enhancing the beauty, drama and suspense for the big thrills at the end. Kudos to director James Cameron and the Skywalker Sound mix team for their restraint and incredible use of dynamic range. That's where the excitement lies for me.

Compressors as Tools To Manipulate Dynamic Range
Compression is a tool; when used by skilled hands, it has produced some of the most beautiful recordings in the world.A lot of music genres are based on the sound of compression, from Disco to Rap to Heavy Metal. And a skilled engineer may intentionally use creative compression to paint a mix and form new special effects; this intended distortion has been used in every style of modern music. This is analogous to the work of the greatest visual artists; many painters are quite capable of producing a natural-looking landscape, but have abandoned that medium to create abstractions which at first glance look like the fingerpaint work of a six-year-old. But a skilled observer realizes what the master artist is communicating. The keys here are intent and skill. Too often, in music, unskilled compression can squash sound, remove the life, vigor and impact, and replace it with boring mush. Many engineers don't know what uncompressed, natural-sounding audio sounds like. It actually takes more work and skill to make a natural-sounding recording than an artificial one. In audio as in the visual arts, first learn to paint naturally; then and only then can you truly understand the art of creating distortion. Learn where compression is useful, and where it does a disservice to the music. A compressed production may sound good on a boombox, but when reproduced on a high-fidelity system, it can sound overbearing and ultimately lifeless. That's why we may need to separately mix "single" and album cuts.

Compressors are commonly used in recording (tracking), mixdown, and in mastering. Everyone has his own style of working with compressors and there are no rules. However, before you make your rules, start by working without any compressors! This learning process will teach you to make better-sounding music later on; the compressor becomes a tool to handle problems, not a crutch or substitute for good recording techniques. First, learn about the natural dynamics and impact of musical instruments, then begin to alter them with compressors (which can include using compression to create special effects). Every 5 years or so, give yourself a reality check...try making a recording or mix with little or no compression. You'll rediscover what I call the microdynamics of music. It's a real challenge, but a refresher course may point out that less compression will buy you a more open, more musical sound than you've previously been getting.

Watch For These Compression-related Pitfalls

Tracking
When tracking vocalists (who have a habit of belting now and then), a well-adjusted compressor can sound reasonably transparent, and most engineers agree the cure is better than the disease. But watch out for a "closed-in" sound, "clamping down" when the vocalist gets loud, or loss of clarity or transparency. Compare IN versus BYPASS before committing to tape. Match levels to make a fair comparison. If you notice too much degradation, maybe it's time to consider a different compressor or change the settings you are using. The sound should be open and clear... remember that no amount of equalization in the mixdown can substitutefor capturing a clear sound quality in the first place. This is true for all the lead instruments, including trumpets and electric guitars. If possible, put the uncompressed sound on a spare track--it may save your life. If there's any "rule," most engineers would agree to save the decision on drum and percussion compression until mixing. There are always exceptions--every piece of music is unique. Just remember, you cannot undo the damage of overcompression, so be careful about compression during tracking.

Mixdown
There are two possible places to apply compressors during mixdown. The first is on individual instruments or stereo pairs; the other is on the console mix buss.

For individual tracks: The same precautions apply to the use of compressors in mixing as in tracking. Start fresh each time--free yourself of preconceptions. Although you compressed the bass on 9 out of the last 10 albums, maybe this time you won't need a compressor. Each musician is an individual. In general, the better the bass player, the less compression you will need to use, and the greater the chance that compression will "choke up" his sound. Get to know the sound of your instrumentalists. What is your mixing philosophy? Are you trying to capture the sound of your instrumentalists or intentionally creating a new sound? While more and more music is created in the control room, it's good practice to know the real sound of instruments; learn how to capture natural sound before moving into the abstract.

In pop music, compressors are often used to create a tighter band sound, making the rhythm instruments sit in a good, constant place in the mix. But when misused or overused, compressors can take away the sense of natural breathing and openness that makes music swing and sway. Thus, I recommend that during mixing, after you've inserted a few compressors on certain instruments (e.g., the bass, rhythm guitar, vocal) and listened for a while, try comparing with the compressors bypassed (automation makes that process easy; store two fader snapshots so you can switch between them). Many times you'll find the compression was hurting the mix, and not helping the sound, by losing the subtleties of the musician's performance. Learn the negative as well as the positive effects of compressors by proving to yourself that you really needed the compressor, or that degree or type of compression. The process of refining a mix should always include revisiting your compression (and EQ) settings and questioning your work. Most music these days is recorded in overdubbed sections, but some performances are still captured at once. An engineer once told me that the best sound he got was the monitor mix on the recording day. By the time he got through slicing and dicing and remixing, all the life was taken out of it (what I call the loss of microdynamics). So remember the sound you got during the recording...did you lose the magic in the mix?

Avoid Wimpy Loud Sound
Some of you may say that my conservative advice only applies to acoustic genres like country music or jazz. However, Rock and Roll music is often a casualty of compressor abuse. I receive rock mixes from well-meaning engineers that should be getting louder and louder and reach a climax, but which have lost their intensity, producing wimpy loud sound. The dynamics of choruses and verses are reversed. Instead of the chorus sounding lively and dramatic, it's been pulled back. When you go against the natural dynamics of the music, the results are less pleasant, and less exciting. I strive to put that kind of life back into sound during the mastering, and my clients are delighted by the results. You can make the mastering engineer's job easier. When you mix rock, listen closely to the climaxes; is it possible that you are killing the music with your compressor? This is a very common problem and only the most skilled mix engineers are able to overcome it, maintaining excitement all the way to the highest peaks. Many mix engineers have trouble handling the duality of rock; they find compressors give them power at mid levels, but they have trouble handling the climaxes; they want them to be loud, but can't seem to do it without overload or overcompression. If you're having those kinds of troubles, don't despair. mastering engineers dislike getting squashed material, because the damage is really hard to fix (though some of the tools I apply are pretty darn effective). Better to send material that's mixed well and powerfully at the mid levels but at the high levels is not squashed. Even if the climaxes don't sound loud enough, consider it a "work in progress." Let the mastering engineer take it to the next level of performance. Using specialized and unique tools, I can remix your material, giving it the punch it needs at mid levels and strength and volume at high levels.

Buss processors
Let me be a bit dogmatic. Reserve "buss compression" or "overall compression" for the mastering stage. As a mastering engineer, I can unequivocally say that the most frequently abused compressor is in the console mix buss. Lately I've been receiving a lot of mixes that have been squashed to death by unintentional misuse or overuse of buss compression.

Overuse of buss compression
Properly-used buss compressors can make music sound louder and more powerful, possibly without deteriorating its character, but is the console mix buss the right place to be working on the loudness character of your music? Absolutely not. If you already have a great-sounding mix without buss compression, then don't add buss compression just to "beef it up." More often than not, the buss compressor you have available will take away the life of your music. Turn up the monitor level if the music doesn't sound loud enough! By all means, leave questions of loudness character out of the mixing process and save them for the mastering stage, where they can be dealt with correctly and effectively (more about that in a moment).

Recently a potential client told me that he was using a little bus compression on his mix. I asked him why he was doing that. He said, "because I think the levels are a little too low." Please don't compress for that reason; if the "levels are too low," then turn them up! The only possible reason to bus compress during a mix is because "it sounds better" to you. I hope that in this article I have provided some useful ways of how you can judge that the mix really "sounds better" before you overall compress. mixing "to the compressor" is also a bit like cheating. Your whole judgment becomes geared to what the compressor is doing rather than the act of mixing itself. When in doubt (and even when not in doubt), mix two versions, one with and one without bus compression, and send both to the mastering house. You may be surprised which version the mastering engineer chooses, and which one sounds better after mastering. Also remember, that not all compressors sound that good. The mastering house might be able to employ a digital compressor like the Weiss, which uses 40 bit floating point internal processing, double sampling, has extraordinary attack and release time flexibility. Or an analog compressor, like the Manley Vari-Mu. Both of these are examples of specialized mastering compressors with extraordinary sound.

One possible proper use of a buss compressor is to "tighten" a mix when individual compressors couldn't do the job. However, be careful not to squash the mix. Tuning a buss compressor is an art born of technical knowledge and experience. As always, compare IN and OUT very carefully, and don't be afraid to patch it OUT if that sounds better. Buss compression causes all the instruments to be modulated by the attack and transients of the loudest instrument. A rim shot or cymbal crash can take down the reverberation and the sound of all the other instruments. There are very few console compressors that are capable of doing buss compression without screwing up transparency, transient response or musical dynamics. Excellent circuit design is required, as well as attack and release characteristics idealized for the job of buss compression. Very few outboard compressors can handle that job. If you want to tighten the mix, first try using submix compression on the rhythm section alone. That way you won't abuse the clarity of the drums and vocal.

Stop Emulating Squashed CDs
Many mixing engineers compare their mixes against already-pressed CDs, but becareful what you choose as a standard. Ironically, mastered CDs often do not sound like what comes out of the mix, so how can you emulate something which can only be done post-mix? What you really need is to hear the sound of a good mix before it was sent for mastering. But since that's not available, choose from the plenitude of pop records that have been well-mixed and mastered, as listed in the CD Honor Roll. When choosing a reference album, don't pick it because it's "hotter" than everyone else; instead, listen for impact, clarity, transparency, ambience, warmth, space, depth, beauty, openness, naturalness, and (sometimes) punch. But "punch" is an ambiguous term; any so-called "mastering engineer" with a $2000 processor can give an album a kind of "punchy" sound, but often sacrificing all the other character that makes music worth listening to. Remember this: when two CDs are presented at equal loudness, nine out of ten musicians prefer the sound of an uncompressed presentation to a compressed one. For the first few seconds, a louder presentation may grab you, but relentless sound quickly becomes fatiguing. Many of today's compact discs have already exceeded the loudness limit--the level above which the sound quality goes down hill while the sound "quantity" goes up. You can't get something for nothing.

If You Can't Make It Sound Good, Make it Loud?
Contrary to some people's beliefs, mastering is not supposed to be the process of making a record hotter than the competition. mastering should be the process of making a record better than the competition. Currently there is a lot of pressure on mastering engineers to make a record hotter than its neighbor. I'm really surprised that more recording engineers are not up in arms about how mastering engineers (on producer's orders?) are ruining their recordings. I'm quite flattered that one recording engineer called me the first mastering engineer to make his recordings sound better. If you're a mastering engineer, wear the red badge of courage; strive for good sound, even if you have to sacrifice a few dB of loudness.

Clients often complain when they have to turn the volume control up when switching CDs. Why aren't they complaining that they have to turn the volume down when a hot disc comes on? One client told me that she loved the sound of her master, but her test CDR was not hot enough when played in rotation on the CD changer in a local bar. This upset me, because it turned out she was comparing her CD against rather trashy-sounding competition and I didn't want to trash up her sound. In the bar, you can't tell the difference in sound quality, but we're making CDs to sound good at home. I told her that she would sacrifice quality if I made her CD any hotter. Ironically, it was already a hot CD, by the standards of last year and the year before, but obviously not this year! Oh, by the way, eventually I compromised, using my best skills to raise the volume on her CD slightly without sacrificing too much in sound quality, but it saddened me that this had to occur. Her CD would have sounded better if it were not as compressed.

The PARTY Button
CD changers present a real problem in client education. I had to tailor the apparent loudness of this client's CD to work with 5 other hand-picked CDs, but I could pick dozens more that are much louder or softer than hers. We have to teach clients that CDs will always differ in volume and that a CD changer is not a radio station. The restaurant CD changer really needs a live DJ, but that's not practical. The solution: put a compressor in the restaurant, the car, and the jukebox, and reserve quality listening for the home, without compression. We should lobby manufacturers to put a compressor button on receivers and future DVD players; with DSP technology this is a snap. They could label it the Party Button! There should be three compression settings--for background listening at low levels, for moderate listening in a noisy environment, and for parties, where you want to switch discs without adjusting the volume control. Panasonic and Sony will sell a million of them, and we engineers will be eternally grateful! The button may be misused by ignorant consumers, but no more than the loudness button I find permanently pushed in 6 out of 10 homes.

Save it for the Mastering
When you're through mixing, your recording is a diamond in the rough, it's not supposed to sound like a "record" until it is mastered. Just make sure the mixes sound great and wait for mastering to add any post-mix processes. When you make copies for your clients, if they have any problems, tell them to turn up their monitors and wait for the mastering. Don't be tempted to use so-called "mastering processors" ("maximizers") before the mastering begins. As mastering engineers, we want to receive the cleanest, highest resolution, unprocessed, original mix tape or disc. In mastering, the individual songs will be levelled (not "normalized"), and elements of your music defined and clarified, turning your record into a work of art. The mastering engineer objectively looks at every song in context in a controlled, familiar acoustic environment, using superior tools, monitoring, experience, and artistry.

The Vicious Circle of Loudness Envy
The practice of overcompression is part of a vicious circle of loudness envy. Sadly, the current crop of CDs is louder and even more squashed than its predecessors because few people have stood up to fight the problem. Participants in this unwitting vicious circle include mix engineers, musicians, producers, mastering engineers, and radio program directors, but the problem is introduced during the mastering process. Many people blame the program director for the problem, but I think we're all partly at fault. Regardless of the cause, we all have to participate in a solution before our music turns into mush.

Program directors should realize that the sound on their office CD player has little to do with the disc's on-air quality. PD's may think the loudest record they hear is the best, but they forget that when it gets to the air, on-air processors will squash it (drop the volume) more than other records. Producers are afraid that the PD will reject their record if they have to turn up the volume. But by now, hot CDs have put the PD's volume control at the bottom of its travel, so where do we go from here? Well, let's get the program directors to make decisions on the merits of the music, not on its loudness character. One way to solve that is to install a compressor in the PD's audition system, one that'll squash music as much as his radio station does. We could call it The Ecumenical Button. Send me suggestions to see how we can get this one done (no kidnapping, please).

From The Sublime to the Ridiculous
Producers don't seem to like making a CD that's even a little softer than the competition, so each succeeding CD is often a little bit hotter. Just how much hotter can CDs get? I can cut a CD that's 16 dB louder than the ones we made in the early 90's, before digital limiters became popular, but it'll look like a square wave and soundlike audio cream of wheat! Imagine the consumer problems caused by large variations in loudness--switching CDs has literally become Russian Roulette, shooting out our speakers and ears! But ultimately, your hot CD doesn't get any louder for the public; they just turn their monitor down, and scream in disgust at the increasing range they have to move the knob when they change CDs. In addition, sound quality is suffering by an unjustifiable demand for hotter CDs. A fellow mastering engineer reminds me that in the early days of CDs, we didn't have any pressure to make them hotter (because there was little competition), and early pop CDs had good, open sound. They're much softer than current CDs, but if you turn up your volume control you'll see their dynamics are much better-sounding. Why do we have to go backwards in sound quality? We mustn't repeat this mistake with the DVD. Part II of my article on Levels discusses 21st century solutions to this problem.

Let's review the basics. The loudness war may have begun with analog records, but the current problem is many decibels worse than it was in analog. LPs were mixed largely with VU meters, which created a degree of monitoring consistency, but today's peak level meters give entirely too much more room for mischief, and today's digital limiters provide the tools to do the mischief. The net result: great consistency problems in CD level. The peak meter is currently being seriously misused. Remember that the upper ranges of the peak meter were designed for headroom, not for level. A compressed piece of music peaking to -6 dBFS can sound much louder than an uncompressed work peaking to 0. mixing and mastering engineers, use compression for creative purposes, but why not master the CD at a lower peak level, and monitor at the same gain you used for your last CD? There's no reason to fill up all those bits if the CD sounds loud enough. Or, useless limiting if you insist on peaking to 0 dBFS. Too many producers are unskilled meter readers; they seem to need all those lights flashing. Try working at a fixed monitor level, with the meter hidden from view. It'll be a very educational experience.

It's come to the point where mastering engineers should think about working differently. In Part II of my article on Levels, I discuss the 21st century solution to this problem, because the future of our DVD Audio is at stake. Only education can stop this vicious spiral. It's time to fight for quality, not quantity. Sound lower than XYZ hit record? Turn up the monitor!

Part III
Tools To Help Keep Us From Overcompressing

How can you tell when proper compression is becoming overcompression? If you don't have good monitors, it's not easy to know when you've crossed the line. The first sign that you're probably going too far is if you start playing with the compressors simply to achieve overall program "loudness" rather than to help you make a great sound. Remember, the mix room is for mixing. If you know you've got a great sound when the monitor is turned up, then all is ok, the mastering engineer can do the rest. The second sign you're probably overcompressing is if you find you're leaning too much on the compressors to make your mix. A program shouldn't mix itself. It takes a lot of work to mix, and depending on the compressors to do that work for you will probably result in a squashed, lifeless sound.

Here are some practical tools you can use to make better-sounding recordings to send to the mastering studio.

1) As mentioned above, install a Dolby-level-calibrated monitor control, connected via a single, high-quality D/A converter. Visit Part II of my article on Levels for more details.

2) Metering. Meters with combined peak and average readings are some of the best protections against overcompression. The Dorrough meter is a good example, as are meters from DK and Pinguin. Part II of my article on Levels discusses how to make best use of these meters. In summary, when mastering, try to keep the "average level" on this meter's average scale from exceeding 0, with occasional "high average levels" to +3 (equivalent to +3 on a VU meter). If you do, you will have obtained approximately a 14dB peak to average ratio. (Peak to average ratio is also known as Crest Factor). The meter is also a good visual aid for visiting producers.

3) Monitoring. A clean, high-headroom monitor system is essential. If your monitor speakers or amplifier saturate, how can you possibly tell if your material is saturating?

I'd like to thank all my friends on the Sonic Solutions Maillist, The Mastering Webboard,the Pro Audio List,

and many of my fellow mastering engineers, for support and ideas. We've been preaching to the converted. Now it's time to transmit these points to the rest of the world.

For further reference, see my articles:
Mastering engineers discuss compression in mixing

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Master the 2-channel art first Everyone is talking about multichannel sound. I have no doubt that well-engineered multi-channel recordings will produce a more natural soundfield than we've been able to achieve in our 2-channel recordings, but it amazes me how few engineers really know how to take advantage of good ol' fashioned 2-channel stereo. I've been making "naturalistic" 2-channel recordings for many years, and there are others working in the pop field who produce 2-channel (pop, jazz, even rock) recordings with beautiful depth and space. I'm rather disappointed in the sound of 2-channel recordings made by simple "pan-potted mono", the typical sound of a rock mix. But it doesn't have to be, if you study the works of the masters.

I wonder if the recording engineers who are disappointed in 2-channel recording may simply be using the wrong techniques. Pan-potted mono techniques,coupled by artificial reverberation--tend to produce a vague, undefined image, and I can understand why many engineers complain about how difficult it is to get definition working in only two channels. They say that when they move to multichannel mixing (e.g., 5.1) that they have a much easier time of it. Granted, though I suggest that first they study how to make a good 2-channel mixdown with depth, space, clarity, and definition. It's possible if you know the tricks. Most of those tricks involve the use of the Haas effect, phase delays, more natural reverbs and unmasking techniques. If engineers don't study the art of creating good 2-channel recordings, when we move to 5.1, ultimately we will end up with more humdrum mixes, more "pan-potted mono", only with more speakers. This article describes techniques that will help you with 2-channel and multichannel recordings. Furthermore, well-engineered 2-channel recordings have encoded ambience information which can be extracted to multichannel, and it pays to learn about these techniques.

The Perception of Depth
At first thought, it may seem that depth in a recording is achieved by increasing the ratio of reverberant to direct sound. But it is a much more involved process. Our binaural hearing apparatus is largely responsible for the perception of depth. But recording engineers were concerned with achieving depth even in the days of monophonic sound. In the monophonic days, many halls for orchestral recording were deader than those of today. Why do monophonic recording and dead rooms seem to go well together? The answer is involved in two principles that work hand in hand: 1) The masking principle and 2) The Haas effect.

The Masking Principle
The masking principle says that a louder sound will tend to cover (mask) a softer sound, especially if the two sounds lie in the same frequency range. If these two sounds happen to be the direct sound from a musical instrument and the reverberation from that instrument, then the initial reverberation can appear to be covered by the direct sound. When the direct sound ceases, the reverberant hangover is finally perceived.

In concert halls, our two ears sense reverberation as coming diffusely from all around us, and the direct sound as having a distinct single location. Thus, in halls, the masking effect is somewhat reduced by the ears' ability to sense direction.

In monophonic recording, the reverberation is reproduced from the same source speaker as the direct sound, and so we may perceive the room as deader than it really is, because of directional masking. Furthermore, if we choose a recording hall that is very live, then the reverberation will tend to intrude on our perception of the direct sound, since both will be reproduced from the same location--the single speaker. So there is a limit to how much reverberation can be used in mono.

This is one explanation for the incompatibility of many stereophonic recordings with monophonic reproduction. The larger amount of reverberation tolerable in stereo becomes less acceptable in mono due to directional masking. As we extend our recording techniques to 2-channel (and eventually multichannel) we can overcome directional masking by spreading reverberation spatially away from the direct source, achieving both a clear (intelligible) and warm recording at the same time.

The Haas Effect
The Haas effect can be used to overcome directional masking. Haas says that, in general, echoes occurring within approximately 40ms of the direct sound become fused with the direct sound. We say that the echo becomes "one" with the direct sound, and only a loudness enhancement occurs.

A very important corollary to the Haas effect says that fusion (and loudness enhancement) will occur even if the closely-timed echo comes from a different direction than the original source. However, the brain will continue to recognize (binaurally) the location of the original sound as the proper direction of the source. The Haas effect allows nearby echoes (up to approximately 40ms delay, typically 30ms) to enhance an original sound without confusing its directionality. We can take advantage of the Haas effect to naturally and effectively convert an existing 2-channel recording to a 4-channel or surround medium. When remixing, place a discrete delay in the surround speakers to enhance and extract the original ambience from a previously recorded source! No artificial reverberator is needed if there is sufficient reverberation in the original source. Here's how it works:

Because of the Haas effect, the ear fuses the delayed with the original sound, and still perceives the direct sound as coming from the front speakers. But this does not apply to ambience--ambience will be spread, diffused between the location of the original sound and the delay (in the surround speakers). Thus, the Haas effect only works for correlated material; uncorrelated material (such as natural reverberation) is extracted, enhanced, and spread directionally. Dolby laboratories calls this effect "the magic surround," for they discovered that natural reverberation was extracted to the rear speakers when a delay was applied to them. Dolby also uses an L minus R matrix to further enhance the separation. The wider the bandwidth of the surround system and the more diffuse its character, the more effective the psychoacoustic extraction of ambience to the surround speakers.

There's more to Haas than this simple explanation. To become proficient in using Haas in mixing, study the original papers on the various fusion effects at different delay and amplitude ratios.

Haas' Relationship to Natural Environments
We may say that the shorter echoes which occur in a natural environment (from nearby wall and floor) are correlated with the original sound, as they have a direct relationship. The longer reverberation is uncorrelated; it is what we call the ambience of a room. Most dead recording studios have little or no ambient field, and the deadest studios have only a few perceptible early reflections to support and enhance the original sound.

In a good stereo recording, the early correlated room reflections are captured with their correct placement; they support the original sound, help us locate the sound source as to distance and do not interfere with left-right orientation. The later uncorrelated reflections, which we call reverberation, naturally contribute to the perception of distance, but because they are uncorrelated with the original source the reverberation does not help us locate the original source in space. This fact explains why the multitrack mixing engineer discovers that adding artificial reverberation to a dry, single-miked instrument may deteriorate the sense of location of that instrument. If the recording engineer uses stereophonic miking techniques and a liver room instead, capturing early reflections on two tracks of the multitrack, the remix engineer will need less artificial reverberation and what little he adds can be done convincingly.

Using Frequency Response to Simulate Depth
Another contributor to the sense of distance in a natural acoustic environment is the absorption qualities of air. As the distance from a sound source increases, the apparent high frequency response is reduced. This provides another tool which the recording engineer can use to simulate distance, as our ears have been trained to associate distance with high-frequency rolloff. An interesting experiment is to alter a treble control while playing back a good orchestral recording. Notice how the apparent front-to-back depth of the orchestra changes considerably as you manipulate the high frequencies.

Recording Techniques to Achieve Front-to-Back Depth Minimalist Techniques

Balancing the Orchestra
A musical group is shown in a hall cross section. Various microphone positions are indicated by letters A-F.

Microphones A are located very close to the front of the orchestra. As a result, the ratio of A's distance from the back compared to the front is very large. Consequently, the front of the orchestra will be much louder in comparison to the rear. Front-to-back balance will be exaggerated. However, there is much to be said in favor of mike position A, since the conductor usually stands there, and he purposely places the softer instruments (strings) in the front, and the louder (brass and percussion) in the back, somewhat compensating for the level discrepancy due to location. Also, the radiation characteristics of the horns of trumpets and trombones help them to overcome distance. These instruments frequently sound closer than other instruments located at the same physical distance because the focus of the horn increases direct to reflected ratio. Notice that orchestral brass often seem much closer than the percussion, though they are placed at similar distances. You should take these factors into account when arranging an ensemble for recording. Clearly, we also perceive depth by the larger ratio of reflected to direct sound for the back instruments.

The further back we move in the hall, the smaller the ratio of back-to-front distance, and the front instruments have less advantage over the rear. At position B, the brass and percussion are only two times the distance from the mikes as the strings. This (according to theory) makes the back of the orchestra 6 dB down compared to the front, but much less than 6 dB in a reverberant hall, because level changes less with distance.

For example, in position C, the microphones are beyond the critical distance--the point where direct and reverberant sound are equal. If the front of the orchestra seems too loud at B, position C will not solve the problem; it will have similar front-back balance but be more buried in reverberation.

Using Microphone Height To Control Depth And Reverberation
Changing the microphone's height allows us to alter the front-to-back perspective independently of reverberation. Position D has no front-to-back depth, since the mikes are directly over the center of the orchestra. Position E is the same distance from the orchestra as A, but being much higher, the relative back-to-front ratio is much less. At E we may find the ideal depth perspective and a good level balance between the front and rear instruments. If even less front-to-back depth is desired, then F may be the solution, although with more overall reverberation and at a greater distance. Or we can try a position higher than E, with less reverb than F.

Directivity of Musical Instruments
Frequently, the higher up we move, the more high frequencies we perceive, especially from the strings. This is because the high frequencies of many instruments (particularly violins and violas) radiate upward rather than forward. The high frequency factor adds more complexity to the problem, since it has been noted that treble response affects the apparent distance of a source. Note that when the mike moves past the critical distance inthe hall, we may not hear significant changes in high frequency response when height is changed.

The recording engineer should be aware of how all the above factors affect the depth picture so he can make an intelligent decision on the mike position to try next. The difference between a B+ recording and an A+ recording can be a matter of inches. Hopefully you will recognize the right position when you've found it.

Beyond Minimalist Recording
The engineer/producer often desires additional warmth, ambience, or distance after finding the mike position that achieves the perfect instrumental balance. In this case, moving the mikes back into the reverberant field cannot be the solution. Another call for increased ambience is when the hall is a bit dry. In either case, trucking the entire ensemble to another hall may be tempting, but is not always the most practical solution.

The minimalist approach is to change the microphone pattern(s) to less directional (e.g., omni or figure-8). But this can get complex, as each pattern demands its own spacing and angle. Simplistically speaking, with a constant distance, changing the microphone pattern affects direct to reverberant ratio.

Perhaps the easiest solution is to add ambience mikes. If you know the principles of acoustic phase cancellation, adding more mikes is theoretically a sin. However, acoustic phase cancllation does not occur when the extra mikes are placed purely in the reverberant field, for the reverberant field is uncorrelated with the direct sound. The problem, of course, is knowing when the mikes are deep enough in the reverberant field. Proper application of the 3 to 1 rule will minimize acoustic phase cancellation. So will careful listening. The ambience mikes should be back far enough in the hall, and the hallmust be sufficiently reverberant so that when these mikes are mixed into the program, no deterioration in the direct frequency response is heard, just an added warmth and increased reverberation. Sometimes halls are so dry that there is distinct, correlated sound even at the back, and ambience mikes would cause a comb filter effect.

Assuming the added ambience consists of uncorrelated reverberation, then theoretically an artificial reverberation chamber should accomplish similar results to those obtained with ambience microphones. The answer is a qualified yes, assuming the artificial reverberation chamber sounds very good and consonant with the sound of the original recording hall.

What happens to the depth and distance picture of the orchestra as the ambience is added? In general, the front-to-back depth of the orchestra remains the same or increases minimally, but the apparent overall distance increases as more reverberation is mixed in. The change in depth may not be linear for the whole orchestra since the instruments with more dominant high frequencies may seem to remain closer even with added reverberation.

The Influence of Hall Characteristics on Recorded Front-to-Back Depth in Live Halls
In general, the more reverberant the hall, the further back the rear of the orchestra will seem, given a fixed microphone distance. In one problem hall the reverberation is much greater in the upper bass frequency region, particularly around 150 to 300 Hz.

A string quartet usually places the cello in the back. Since that instrument is very rich in the upper bass region, in this problem hall the cello always sounds further away from the mikes than the second violin, which is located at his right. Strangely enough, a concertgoer in this hall does not notice the extra sonic distance because his strong visual sense locates the cello easily and does not allow him to notice an incongruity. When he closes his eyes, however, the astute listener notices that, yes, the cello sounds further back than it looks!

It is therefore rather difficult to get a proper depth picture with a pair of microphones in this problem hall. Depth seems to increase almost exponentially when low frequency instruments are placed only a few feet away. It is especially difficult to record a piano quintet in this hall because the low end of the piano excites the room and seems hard to locate spatially. The problem is aggravated when the piano is on half-stick, cutting down the high frequency definition of the instrument.

The miking solution I choose for this problem is a compromise; close mike the piano, and mix this with a panning position identical to the piano's virtual image arriving from the main mike pair. I can only add a small portion of this close mike before the apparent level of the piano is taken above the balance a listener would hear in the hall. The close mike helps solidify the image and locate the piano. It gives the listener a little more direct sound on which to focus.

Very Dead Rooms
Can minimalist techniques work in a dead studio? Not very well. My observations are that simple miking has no advantage over multiple miking in a deadroom. I once recorded a horn overdub in a dead room, with six tracks of close mikes and two for a more distant stereo pair. In this dead room there were no significant differences between the sound of this "minimalist" pair and six multiple mono close up mikes! The close mikes were, of course, carefully equalized, leveled and panned from left to right. This was a surprising discovery, and it points out the importance of good hall acoustics on a musical sound. In other words, when there are no significant early reflections, you might as well choose multiple miking, with its attendant post-production balance advantages.

Miking Techniques and the Depth Picture
The various simple miking techniques reveal depth to greater or lesser degree. Microphone patterns which have out of phase lobes (e.g., hypercardioid and figure-8) can produce an uncanny holographic quality when used in properly angled pairs. Even tightly-spaced (coincident) figure-8's can give as much of a depth picture as spaced omnis. But coincident miking reduces time ambiguity between left and right channels, and sometimes we seek that very ambiguity. Thus, there is no single ideal minimalist technique for good depth, and you should become familiar with the relative effects on depth caused by changing mike spacing, patterns, and angles. For example, with any given mike pattern, the farther apart the microphones of a pair, the wider the stereo image of the ensemble. Instruments near the sides tend to pull more left or right. Center instruments tend to get wider and more diffuse in their image picture, harder to locate or focus spatially.

The technical reasons for this are tied in to the Haas effect for delays of under approximately 5ms. vs. significantly longer delays. With very short delays between two spatially located sources, the image location becomes ambiguous. A listener can experiment with this effect by mistuning the azimuth on an analog two-track machine and playing a mono tape over a well-focused stereo speaker system. When the azimuth is correct, the center image will be tight and defined. When the azimuth is mistuned, the center image will get wider and acoustically out of focus. Similar problems can (and do) occur with the mike-to-mike time delays always present in spaced-pair techniques.

The Front-to-back Picture with Spaced Microphones
I have found that when spaced mike pairs are used, the depth picture also appears to increase, especially in the center. For example, the front line of a chorus will no longer seem straight. Instead, it appears to be on an arc bowing away from the listener in the middle. If soloists are placed at the left and right sides of this chorus instead of in the middle, a rather pleasant and workable artificial depth effect will occur. Therefore, do not overrule the use of spaced-pair techniques. Adding a third omnidirectional mike in the center of two other omni's can stabilize the center image, and proportionally reduces center depth.

Multiple Miking Techniques
I have described how multiple close mikes destroy the depth picture; in general I stand behind that statement. But soloists do exist in orchestras, and for many reasons, they are not always positioned in front of the group. When looking for a natural depth picture, try to move the soloists closer instead of adding additional mikes, which can cause acoustic phase cancellation. But when the soloist cannot be moved, plays too softly, or when hall acoustics make him sound too far back, then a close mike or mikes (known as spotmikes) must be added. When the close solo mikes are a properly placed stereo pair and the hall is not too dead, the depth image will seem more natural than one obtained with a single solo mike.

Apply the 3 to 1 rule. Also, listen closely for frequency response problems when the close mike is mixed in. As noted, the live hall is more forgiving. The close mike (not surprisingly) will appear to bring the solo instrument closer to the listener. If this practice is not overdone, the effect is not a problem as long as musical balance is maintained, and the close mike levels are not changed during the performance. We've all heard recordings made with this disconcerting practice. Trumpets on roller skates?

Delay Mixing
At first thought, adding a delay to the close mike seems attractive. While this delay will synchronize the direct sound of that instrument with the direct sound of that instrument arriving at the front mikes, the single delay line cannot effectively simulate the other delays of the multiple early room reflections surrounding the soloist. The multiple early reflections arrive at the distant mikes and contribute to direction and depth. They do not arrive at the close mike with significant amplitude compared to the direct sound entering the close mike. Therefore, while delay mixing may help, it is not a panacea.

Influence of the Control Room Environment on Perceived Depth
At this point, many engineers may say, "I've never noticed depth in my control room!" The widespread practice of placing near-field monitors on the meter bridges of consoles kills almost all sense of depth. Comb-filtering and sympathetic vibrations from nearby surfaces destroy the perception of delicate time and spatial cues. The recent advent of smaller virtual control surfaces has helped reduce the size of consoles, but seek advice from an expert acoustician if you want to appreciate and manipulate depth in your recordings. We should all do this before we expand to multi-channel, for we still have a lot to learn about taking advantage of the hidden depth in 2-channel recordings.

Musical Examples to Check Out
Check out the CD Honor Roll for examples of fantastic recordings. Standard multitrack music recording techniques make it difficult for engineers to achieve depth in their recordings. Mixdown tricks with reverb and delay may help, but good engineers realize that the best trick is no trick: Record as much as you can by using stereo pairs in a live room. Here are some examples of audiophile records I've recorded that purposely take advantage of depth and space, both foreground and background, on Chesky Records. Sara K. Hobo, Chesky JD155. Check out the percussion on track 3, BrickHouse... Johnny Frigo, Debut of a Legend, Chesky JD119. Check out the sound of the drums and the sax on track 9, I Love Paris. Ana Caram, The Other Side of Jobim, Chesky JD73. Check out the percussion, cello and sax on Correnteza. Carlos Heredia, Gypsy Flamenco, Chesky WO126. Play it loud! And listen to track 1 for the sound of the background singers and handclaps. Phil Woods, Astor and Elis, Chesky JD146, for the natural-sounding combination of intimacy and depth of the jazz ensemble.

Technological Impediments to Capturing Recorded Depth
Depth is the first thing to suffer when low-resolution technology is used. Here is a list of some of the technical practices that when misused, or accumulated, can contribute to a boringly flat, depthless recorded picture: Multitrack and multimike techniques, small/dead recording studios, low resolution recording media, amplitude compression, improper use of dithering, cumulative digital processing, and low-resolution digital processing (e.g., using single-precision as opposed to double or higher-precision equalizers). When recording, mixing and mastering-use the best miking techniques, room acoustics, and highest resolution technology, and you'll resurrect the missing depth in your recordings.

Thanks to:

My assistant, David Holzmann, for transcribing my original 1981 article, which I have herein revised and updated.

Lou Burroughs, whose 1974 book Microphones: Design and Application, now out of print, is still one of the prime references on this subject and covers the topic of acoustic phase cancellation. Burroughs invented the 3-to-1 rule, expressed simply: When a sound source is picked up by one microphone and also "leaking" into another microphone that is mixed to the same channel----make sure the distance between microphones is at least three times the distance of the first mike from the source. This general rule does not account for the intensities of all instruments or all room acoustics, but you should listen critically when your microphone distances decrease. And remember, this applies to mixing to one channel, for non-coincident stereo microphone techniques can break the 3-t-1 rule, and if so, be sure to check the sound in mono for phase cancellation.

E. Roerback Madsen, whose article "Extraction of Ambiance Information from Ordinary Recordings" can be found in the 1970 October issue of the Journal of the Audio Engineering Society. Covers the Haas effect and its correlary.

Don Davis, who first defined "critical distance" and many other acoustic terms.

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Use or Build your own Bitscope!

Attention, Sherlock Holmes's of the audio world. You can use an ordinary oscilloscope (20 MHz or better) to see the bit activity of your digital processors, consoles and workstations. Steinberg's Wavelab and many other software programs have a bitscope meter as well. Once you install it, you'll find the bitscope is as essential in the modern-day digital studio as a phase meter. To learn more about how the bitscope has saved the day in studios, read our article, More Bits Please. These photos illustrate some typical bitscope displays.

How to Build a Bitscope
Every digital audio recorder, processor or console extracts serial DATA and WORDCLOCK from the AES/EBU or S/PDIF line. Pick a "neutral" machine or processor that you can patch into your digital audio system at the end of your processing or monitoring chain, so you can analyze what all your processors are doing to the signal. All you have to do is connect the vertical input of your oscilloscope to DATA, and its trigger or timebase to WORDCLOCK (44.1 or 48 KHz), to see which bits and how many bits are being used at all times. If you're not used to digging into audio equipment, then give the job to someone who is. Opening any manufacturer's gear may void the warranty.

Crystal Semiconductor's ubiquitous CS8412 digital receiver IC is used in many processors. You'll find DATA on pin 26, and WORDCLOCK on pin 11 of this 28-pin chip. Attach the shield of the scope lines to ground. I suggest soldering a 75 ohm build-out (isolation) resistor from the chip's pins to the scope lines, to protect the signals from accidental shorts. Use good, short coax cables (I've used three feet with no problems).

You can still be a digital detective even if you're not the do-it-yourself type. Digital Domain will add scope outputs to its FCN-1 Format Converter or VSP/P Digital Audio Control Center for a small fee. For further information, contact Digital Domain 1-800-DIGIDO-1 or email us.

Interpreting the Display
The bitscope will tell you when certain things are wrong (e.g., missing bits, or extra bits), but it can't guarantee that everything is right (e.g., harmonic distortion will not show on the bitscope). Use the bitscope as a visual aid, a first line of defense against digital audio problems. Your ears and your knowledge must do the rest.

The 8412 chip can be configured for many modes. The most common mode presents one channel's worth of data on wordclock "up," and the other channel on wordclock "down." Crystal uses a 64-bit "slot," so you'll see up to 24 bits worth of one channel, followed by 8 bits of "silence," then the other channel (another 32-bit half-slot). Counting bits is easy if you adjust your scope's timebase to show one audio channel, and 2-1/2 bits per division, which gives a convenient count of 5 bits every two divisions, and spreads 24 bits across the whole screen. The format is 2's complement, with the MSB at left, and LSB at right. When the MSB is low, the audio signal is positive, when high, it's negative, so the MSB will be toggling all the time, unless the signal is pure DC. A toggling bit will appear to have both high and low values, this just means that the eye's persistence of vision is showing both values.

These scope pictures are a little over-exposed, so the top vertical line is fatter and brighter than the actual scope display.

This is the bitscope, showing one channel, full scale 16-bit sine wave. Note the handwritten scale on the top of the chassis.

16-bit sinewave at -20 dBFS. I have added a computer-driven counter scale to these images to make it easy to identify the bits.

16-bit sinewave at -60 dBFS.

16-bit sinewave at -80 dBFS.

24-bit full scale sinewave.

24-bit sinewave at -50 dBFS.

20-bit full scale sinewave.

"Defective" digital processor in BYPASS. Source is a 16-bit sinewave at -70 dBFS. The additional bits could be DC offset or what?

Defective dithering processor set for 16-bit output. Source is a 16-bit full-scale sinewave. Note the missing bit in the 17th position and an extra 18th bit is toggling.

The same defective dithering processor idling (with no input signal). Note the faint line showing the 14th bit is toggling, along with the 15 and 16th bits, plus the same missing bit at the 17th position, and the toggling 18th bit.

Dithering processor in idle (no input signal), showing 4 bits toggling (high order dither with noise-shaping).

Keeping Your Digital Audio Pure from First Recording to Final Master

Part I

Dither is not the most important technical detail to learn about, but if you want to get your digital audio done just right, then you should learn about dither. Especially if you want to learn why your digital reverb has been leaching the ambience out of your music, when it's supposed to be adding ambience. Or why your CDs don't sound as spacious as your 24 bit sources and want to avoid that veiled, dry, and lifeless feeling!

Follow that Sample

Let's start with a little lesson in DSP (Digital Signal Processors). Many workstation and processor manufacturers ignore the critical issue of wordlength. Let's examine what happens to digital audio when you change gain (or mix, equalize, compress, sample rate convert, or perform any type of calculation) in a digital audio workstation. It's all arithmetic, isn't it? Yes, but the accuracy of that arithmetic, and how you (or the workstation) deal with the arithmetic product, can make the difference between pure-sounding digital audio or digital sand paper.

All DSPs deal with digital audio on a sample by sample basis. At 44.1kHz, there are 44,100 samples in a second (88,200 stereo samples). When changing gain, the DSP looks at the first sample, performs a multiplication, spits out a new number, and then moves on to the next sample. It's that simple.

Instead of losing you with esoteric concepts like two's complement notation, fixed vs. floating point, and other digital details, I'm going to talk about digital dollars. Suppose that the value of your first digital audio sample was expressed in dollars instead of volts, for example, one dollar and fifty one cents--$1.51. And suppose you wanted to take it down (attenuate it) by 6 dB. If you do this wrong, you'll lose more than money, by the way. 6 dB is half the original value (it has to do with logarithms; don't worry about it). So, to attenuate our $1.51 sample, we divide it by 2.

Oops! $1.51 divided by 2 equals 75-1/2 cents, or .755. So, we've just gained an extra decimal place. What should we do with it, anyway? It turns out that dealing with extra places is what good digital audio is all about. If we just drop the extra five, we've theoretically only lost half a penny--but you have to realize that half a penny contains a great deal of the natural ambience, reverberation, decay, warmth, and stereo separation that was present in the original $1.51 sample! Lose the half penny, and there goes your sound. The dilemma of digital audio is that most calculations result in a longer wordlength than you started with. Getting more decimal places in our digital dollars is analogous to having more bits in our digital words. When a gain calculation is performed, the wordlength can increase infinitely, depending on the precision we use in the calculation. A 1 dB gain boost involves multiplying by 1.122018454 (to 9 place accuracy). Multiply $1.51 by 1.122018454, and you get $1.694247866 (try it on your calculator). Every extra decimal place may seem insignificant to you, until you realize that DSPs require repeated calculations to perform filtering, equalization, and compression. 1 dB up here, 1 dB down here, up and down a few times, and the end number may not resemble the right product at all, unless adequate precision is maintained. Remember, the more precision, the cleaner your digital audio will sound in the end (up to a reasonable limit).

The First Secret of Digital Audio
Now you know the first critical secret of digital audio: wordlengths expand. If this concept is so simple, why is it disregarded by some manufacturers? The answer is in your wallet. While DSPs are capable of performing double and triple precision arithmetic (all you have to do is store intermediate products in temporary storage registers), it slows them down, and complicates the whole process. It's a hard choice, entirely up to the DSP programmer/processor designer, who's been put under the gun by management to fit more program features into less space, for less money. Questions of sound quality and quantization distortion can become moot compared to the selling price.

Inside a digital mixing console (or workstation), the mix buss must be much longer than 16 bits, because adding two (or more) 16-bit samples together and multiplying by a coefficient (the level of the master fader is one such coefficient) can result in a 32-bit (or larger) sample, with every little bit significant. Since the AES/EBU standard can carry up to 24-bits, it is practical to take the 32-bit word, round it down to 24 bits, then send the result to the outside world, which could be a 24-bit storage device (oranother processor). The next processor in line may have an internal wordlength of 32 or more bits, but before output it must round the precision back to 24 bits. The result is a slowly cumulating error in the least significant bit(s) from process to process. Fortunately, the least significant bit of a 24-bit word is 144 dB down, and most sane people recognize that degree of error to be inaudible.

Something For Nothing?
But suppose you want to record the digital console's output to a 16 bit medium, like the CD. Frankly, it's a serious compromise to take your console's 24-bit output word and truncate it to 16 bits. After processing, the mastering engineer uses a technique called dithering to take long wordlengths, and cleanly turn them to 16-bit for the CD. First, must ensure that our DAW is high resolution (has very low distortion at low levels) and can be bit-transparent when called upon. Bit-transparent means that the output is identical to the source, from the most significant to the least significant bit, that the DAW does not increase or decrease the source wordlength.

Good Advice
Once you've verified your workstation is bit-transparent, then proceed with editing, with the goal of maintaining the integrity of your original source. Do not change gain unless you need to align the gains of two pieces you are editing together. Do not normalize (normalization is just changing gain). Do not equalize. Do not fade in or fade out. Just edit. This is to avoid additional DSP or degradation when the mix gets to the mastering studio .Leave the segues, fadeouts and gain changes for the mastering house, where they can properly handle the long wordlengths necessary for smooth fades (so that's why your last fadeout sounded like it dropped off a cliff!). Follow these simple guidelines and your digital audio will immediately start sounding better.

Part II

Dither

How to Dither Let's look at that long sample word. Whether it's 24 bits or 32 bits, we have to find some way to move the important information contained in the lower (least significant) bits into the upper 16 bits for recording to the CD standard. Truncation is very bad. What about rounding? In our digital dollar example, we ended up with an extra 1/2 cent. In grammar school, they taught us to round the numbers up or down according to a rule (we learned "even numbers...roundup, odd...round down"). But when we're dealing with more numerical precision and small numbers that are significant, it gets a little more complicated.

It turns out the best solution for maintaining the resolution of digital audio is to calculate random numbers and add a different random number to every sample. Then, cut it off at 16 bits. The random numbers must also be different for left and right samples, or else stereo separation will be compromised.

For example:

Starting with a 24-bit word (each bit is either a 1 or a 0 in binary notation):

Upper 16 bits       Lower 8

Original 24-bit Word MXXX XXXX XXXX XXXW YYYY YYYY

Add random number                        ZZZZ ZZZZ

The result of the addition of the Z's with the Y's gets carried over into the new least significant bit of the 16-bit word (LSB, letter W above), and possibly higher bits if you have to carry. In essence, the random number sequence combines with the original lower bit information, modulating the LSB. Therefore, the LSB, from moment to moment, turns on and off at the rate of the original low level musical information. The random number is called dither; the process is called redithering, to distinguish from the original dithering process used to during the original recording.

Random numbers such as these translate to random noise (hiss) when converted to analog. The amplitude of this noise is around 1 LSB, which for 16 bit lies at about 96 dB below full scale. By using dither, ambience and decay in a musical recording can be heard down to about -115 dB, even with a 16-bit wordlength. Thus, although the quantization steps of a 16-bit word can only theoretically encode 96 dB of range, with dither, there is an audible dynamic range of up to 115 dB! The maximum signal-to-noise ratio of a dithered 16-bit recording is about 96 dB. But the dynamic range is far greater, as much as 115 dB, because we can hear music below the noise. Usually, manufacturer's spec sheets don't reflect these important specifications, often mixing up dynamic range and signal-to-noise ratio. Signal-to-noise ratio (of a linear PCM system) is the RMS level of the noise with no signal applied expressed in dB below maximum level (without getting into fancy details such as noise modulation). It should be, ideally, the level of the dither noise. Dynamic range is a subjective judgment more than a measurement--you can compare the dynamic range of two systems empirically with identical listening tests. Apply a 1 kHz tone, and see low you can make it before it is undetectable. You can actually measure the dynamic range of an A/D converter without an FFT analyzer. All you need is an accurate test tone generator and your ears, and a low-noise headphone amplifier with sufficient gain. Listen to the analog output and see when it disappears (use a real good 16 bit D/A for this test). Another important test is to attenuate music in your workstation (about 40 dB) and listen to the output of the system with headphones. Listen for ambience and reverberation; a good system will still reveal ambience, even at that low level. Also listen to the character of the noise--it's a very educating experience.

Some Tests for Linearity
You can verify whether your digital audio workstation truncates digital words or does other nasty things, without any measurement instruments except your ears. Obtain the disc Best of Chesky Classics and Jazz and Audiophile Test Disc, Vol. III, Chesky JD111.* Track 42 is a fade to noise without dither, demonstrating quantization distortion and loss of resolution. Track 43 is a fade to noise with white noise dither, and track 44 uses noise-shaped dither (to be explained). Use Track 43 as your test source; you should be able to hear smooth and distortion-free signal down to about -115 dB. Then listen to track 44 to see how much better it can sound. Try processing track 43 with digital equalization or level changes (both gain and attenuation, with and without dither, if it's available in your workstation) to see what they do to the sound. If your workstation is not up to par, you'll be shocked. Use a quiet, high-gain headphone amplifier to help reveal the low level problems.

*available at major record chains or through Chesky Records, Box 1268, Radio City Station, New York, NY 10101; 212-586-7799. The hard-to-find CBS CD-1, track 20, also contains a fade to noise test.

So Little Noise, So Much Effect
-96 dB seems like so little noise. But strangely, engineers have been able to hear the effect of the dither noise, even at normal listening levels. Dither noise helps us recover ambience, but conversely it also obscures the same ambience we've been trying to recover! Dither noise adds a slight veil to the sound. That's why I say, dither, you can't live with it, and you can't live without it.

Improved Dithering Techniques
Where there's a will, there's a way. Although the required amplitude of the dither is about -96 dB, it's possible to shape (equalize) the dither to minimize its audibility. Noise-shaping techniques re-equalize the spectrum of the dither while retaining its average power, moving the noise away from the areas where the ear is most sensitive (circa 3 KHz), and into the high frequency region (10-22 KHz).

Here is a picture of one of the most successful noise-shaping curves (courtesy of Meridian Audio, Ltd).



As you can see, it is a very high-order filter, requiring considerable calculation, with several dips where human hearing is most sensitive. The sonic result is an incredibly silent background, even on a 16-bit CD. The 0 dB line is around -96 dBFS in this diagram.

There are numerous noise-shaping redithering devices on the market. Very high precision (56 to 72 bit) arithmetic is required to calculate these random numbers. One box uses the resources of an entire DSP chip just to calculate dither. The sonic results of these noise-shaping techniques range from very good to marvelous. The best techniques are virtually inaudible to the ear. With 72-bit arithmetic, all the dither noise has been pushed into the high frequency region, which at -60 or -70 dB is still inaudible. Critical listeners were complaining that the high frequency rise of the early noise-shaping curves changed the tonality of the sound, adding a bit of brightness. But it turns out that it is the shape of the curve in the midband that affects the tonality, due to masking. Two or three of the latest and best of these noise-shaping dithers are tonally neutral, to my ears. It took a long time to get there (about 10 years of development), but now we can say that the best of these processors yield 19-20 bit performance on a 16-bit CD, with virtually no tonal alteration or loss of ambience from the 24-bit source.

Noise-shapers on the market include: db Technologies model 3000 Digital Optimizer, Meridian Model 618, Sony Super Bit Mapping, Waves L1 and L2 Ultramaximizers, Prism, POW-R, and several others. When using dithering plugins, be sure to use them with the right version of workstation software to retain a 24-bit wordlength until the final mastering step.

Apogee Electronics produced the UV-22 system, in response to complaints about the sound of earlier noise-shaping systems, declaring that 16-bit performance is just fine. They do not use the word "dither" (because their noise is periodic, they prefer to call it a "signal"), but it smells like dither to me. Instead of noise-shaping, UV-22 adds a carefully calculated noise at around 22 KHz, without altering the noise in the midband.

To effectively compare the sound and resolution of these redithering techniques, perform the low level test described above. Feed low level 24-bit music (around -40 dB) into the processor, and listen to the output at high gain in a pair of headphones with a good quality D/A converter. You will be shocked to hear the sonic differences between the systems. Some will be grainy, some noisy, and some distorted, indicating improper dithering or poor calculation. The winner of this test should be your choice of dithering processor, although at high gains you are exaggerating the effect of the extra high frequencies in the noise, which would not be noticed at normal gains..

Damage, Destruction, or just Deterioration?
Before digital recording and editing, every edit was destructive. Every equalization or gain change involved an analog copy, with attendant noise, or remixing the multitrack, which "destroys" or replaces the previous mixdown. After DAWs were invented, people started talking about "non-destructive"-editing, and keeping your sound in the digital domain until the end. But as we have seen, even "non-destructive" may be damaging if word lengths aren't maintained.

The Best Approach
To maintain the quality of your digital audio, always store the full output wordlength of your digital processors. Also, be sure to question authority. Never take a digital processor for granted. Don't even trust BYPASS mode, unless you're sure the processor produces true clones in bypass. The following illustration (courtesy of Jim Johnston, AT&T research), shows a series of FFT plots of a sine wave. The top row is an undithered 16 bit sine wave. Note the distortion products (vertical spikes at regular intervals). The second row is that sine wave with uniform dither. Note how the distortion products are now gone. The bottom row is the dithered sine wave, going through a popular model of digital processor set for BYPASS and truncated to 16 bits. This is what would happen if you took your source, fed it through this processor in BYPASS mode, and recorded it again!

Disarming, isn't it? That's why you should arm yourself with a bitscope or test every processor you own for bit transparency before attempting to make master-quality work with those processors patched in your signal chain.

The Cost of Cumulative Dithering
When feeding processors, DAWs or digital mixers to your recording unit, dither the output of the processor to a 24-bit word. Dithering always sounds better than truncation without dither. But to avoid adding a veil to the sound, avoid cumulative dithering, in other words, multiple generations of any dither. Make sure that redithering to 24- or 16-bit is the one-time, final process in your project. For related information visit my article More Bits, Please. When performed properly, dithering will help your music to retain its depth and purity of tone.

December 2007
I hesitate to remove this older article from our website, as it is still informative, but I highly recommend that those interested in the latest word on this subject please read the chapter on jitter in my new book. Some questions that this previous article has raised have been clarified much better in the book. -BK

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Jitter is so misunderstood among recording engineers and audiophiles that we have decided to devote a Section to the topic. All digital devices that have an input and an output can add jitter to the signal path. For example, Digital Domain's FCN-1 Format Converter adds a small amount of jitter (around 200 ps RMS) to the digital audio signal path. Is this good? Is it bad? What sonic difference does it make? We will attempt to answer these--and other important--questions in this Section.

What is Jitter?
Jitter is time-base error. It is caused by varying time delays in the circuit paths from component to component in the signal path. The two most common causes of jitter are poorly-designed Phase Locked Loops (PLL's) and waveform distortion due to mismatched impedances and/or reflections in the signal path.

Here is how waveform distortion can cause time-base distortion:



The top waveform represents a theoretically perfect digital signal. Its value is 101010, occuring at equal slices of time, represented by the equally-spaced dashed vertical lines. When the first waveform passes through long cables of incorrect impedance, or when a source impedance is incorrectly matched at the load, the square wave can become rounded, fast risetimes become slow, also reflections in the cable can cause misinterpretation of the actual zero crossing point of the waveform. The second waveform shows some of the ways the first might change; depending on the severity of the mismatch you might see a triangle wave, a squarewave with ringing, or simply rounded edges. Note that the new transitions (measured at the Zero Line) in the second waveform occur at unequal slices of time. Even so, the numeric interpretation of the second waveform is still 101010! There would have to be very severe waveform distortion for the value of the new waveform to be misinterpreted, which usually shows up as audible errors--clicks or tics in the sound. If you hear tics, then you really have something to worry about.

If the numeric value of the waveform is unchanged, why should we be concerned? Let's rephrase the question: "when (not why) should we become concerned?" The answer is "hardly ever." The only effect of timebase distortion is in the listening; as far as it can be proved, it has no effect on the dubbing of tapes or any digital to digital transfer (as long as the jitter is low enough to permit the data to be read. High jitter may result in clicks or glitches as the circuit cuts in and out). A typical D to A converter derives its system clock (the clock that controls the sample and hold circuit) from the incoming digital signal. If that clock is not stable, then the conversions from digital to analog will not occur at the correct moments in time. The audible effect of this jitter is a possible loss of low level resolution caused by added noise, spurious (phantom) tones, or distortion added to the signal.

A properly dithered 16-bit recording can have over 120 dB of dynamic range; a D to A converter with a jittery clock can deteriorate the audible dynamic range to 100 dB or less, depending on the severity of the jitter. I have performed listening experiments on purist, audiophile-quality musical source material recorded with a 20-bit accurate A/D converter (dithered to 16 bits within the A/D). The sonic results of passing this signal through processors that truncate the signal at -110, -105, or -96 dB are: increased "grain" in the image, instruments losing their sharp edges and focus; reduced soundstage width; apparent loss of level causing the listener to want to turn up the monitor level, even though high level signals are reproduced at unity gain. Contrary to intuition, you can hear these effects without having to turn up the listening volume beyond normal (illustrating that low-level ambience cues are very important to the quality of reproduction). Similar degradation has been observed when jitter is present. Nevertheless, the loss due to jitter is subtle, and primarily audible with the highest-grade audiophile D/A converters.

Jitter And the AES/EBU Interface
The AES/EBU (and S/PDIF) interface carries an embedded clock signal. The designers of the interface did not anticipate that it could cause a subtle amount of jitter due to the nature of the preamble in the AES/EBU signal. The result is a small amount of program-dependent jitter which often sounds like an intermodulation, a high-frequency edge added to the music. To minimize this effect in the listening, use a D/A converter with a high degree of internal jitter reduction. An external jitter reduction device that removes the subcode signal (containing time of day, start IDs, etc.) also helps.

The SDIF-2 (Sony Digital Interface-2) uses a separate cable for the clock signal, and thus is not susceptible to program-dependent jitter. However, the quality of the PLL used to detect an SDIF-2 wordclock is still important to low jitter. It is much easier to build a low-jitter PLL for a wordclock signal than for an AES/EBU signal.

Is Jitter Cumulative? What About My Dubs?
Consider a recording chain consisting of an A to D Converter, followed by the FCN-1, feeding a DAW , and finally a D to A Converter. During the recording, the jitter you will hear is dependent on the ability of the last PLL in the chain (in the D to A) to reduce the cumulative jitter of the preceding elements in the chain. The time-base error in the D to A is a complex aggregate of the timebase errors of all the preceding devices, including their ability to reject incoming jitter, plus the D to A's ability to reject any jitter coming into it. During the recording, there are 3 Phase Locked Loops in the chain: in the FCN-1, the recorder, and the D to A converter. Each PLL has its own characteristics; many good PLLs actually reduce incoming jitter; others have a high residual jitter. It is likely that during playback, you will hear far less jitter (better low level resolution, clearer highs) because there is only one PLL in the digital chain, between the playback deck and the D to A. In other words, the playback will sound better than the sound monitored while recording!

Jitter and A to D Converters
The A to D Converter is one of the most critical digital audio components susceptible to jitter, particularly converters putting out long word lengths (e.g. 24-bits). The master clock that drives an A/D converter must be very stable. A jittery master clock in an A/D converter can cause irrevocable distortion and/or noise which cannot be cancelled out or eliminated at further stages in the chain. A/D's can run on internal or external sync. On internal sync, the A/D is running from a master crystal oscillator. On external sync, the A/D's master clock is driven by a PLL, which is likely to have higher remnant jitter than the crystal clock. That is why I recommend running an A/D converter on internal clock wherever possible, unless you are synchronizing an A/D to video or to another A/D (in a multichannel setup). If you must use external sync, use the most stable external source possible (preferably video or wordclock over AES/EBU), and try to ensure that the A/D's designer used an ultra-stable PLL.

Jitter and DSP-based Processors
Most DSP-based software acts as a "state machine." In other words, the output result on a sample by sample basis is entirely predictable based on a table of values of the incoming samples. The regularity (or irregularity) of the incoming clock has no effect on the output data.

Exceptions to "state-based" DSP processes include Asynchronous Sample Rate Converters, which are able to follow variations in incoming sample rate, and produce a new outgoing sample rate. Such devices are not "state-machines", and jitter on the input may affect the value of the data on the output. I can imagine other DSP processes that use "time" as a variable, but these are so rare that most normal DSP processes (gain changing, equalization, limiting, compression, etcetera) can be considered entirely to be state machines.

Therefore, as far as the integrity of the data is concerned, I have no problems using a chain of jittery (or non-jittery) digital devices to process digital audio, as long as the digital device has a high integrity of DSP coding (passes the "audio transparency" test).

Why are plug-in computer cards so jittery? Does this affect my work with the cards?
Many computer-based digital audio cards have quite high jitter, which makes listening through them a variable experience. It is very difficult to design a computer-based card with a clean clock--due to ground and power contamination and the proximity of other clocks on the computer's motherboard. The listener may leap to a conclusion that a certain DSP-based processor reduces soundstage width and depth, low level resolution, and other symptoms, when in reality the problem is related to a jittery phase-locked loop in the processor input, not to the DSP process itself. Therefore, always make delicate sonic judgments of DSP processors under low jitter conditions, which means placing high-quality jitter reduction units throughout the signal chain, particularly in front of (and within) the D/A converter. Sonic Solutions's USP system has very low jitter because its clocks are created in isolated and well-designed external I/O boxes.

Jitter and Digital Copies
The key is in the playback, not in the transfer
Many well-known devices have high jitter on their outputs, especially DAT machines. However, for most digital to digital transfers, jitter is most likely irrelevant to the final result. I said "most likely" because a good scientist always leaves a little room for doubt in the face of empirical (listening) evidence, and I have discovered certain audible exceptions (see below). Until we are able to measure jitter with widely-available high-resolution measuring equipment, and until we can correlate jitter measurements adequately against sonic results, I will leave some room for doubt.

Playback from a DAT recorder usually sounds better than the recording, because there is less jitter. Remember, a DAT machine on playback puts out numbers from an internal RAM buffer memory, locked to its internal crystal clock. A DAT machine that is recording (from its digital input) is locked to the source via its (relatively jittery) Phase Locked Loop. As the figure above illustrates, the numbers still get recorded correctly on tape, although their timebase was jittery while going in. Nevertheless, on playback, that time base error becomes irrelevant, for the numbers are reclocked by the DAT machine! I have not seen evidence that jitter is cumulative on multiple digital dubs. In fact, a Compact Disc made from a DAT master usually sounds better than the DAT... because a CD usually plays back more stably than a DAT machine. The fact that a dub can sound better than the original is certainly a tough concept to believe, but it is one key to understanding the strange phenomenom called Digital Audio.

It's unnerving to hear a dub that sounds different from the original, so I've performed some tests to try to see if jitter is accumulated. I think I've proved with reasonable satisfaction, that under most conditions jitter is not accumulated on multiple dubs, and that passing jittery sources through a storage medium (such as hard disk) results in a very non-jittery result (e.g., recorded CDR).

Here are two tests I have made (this is far from a complete list):

Test #1
I produced a 99th-generation versus 1st-generation audio test on Chesky Records' first Test CD. If jitter were accumulated on subsequent dubs, then the 99th generation would sound pretty bad, right? Well, most people listening to this CD can't tell the difference and there is room for doubt that there is a difference. It's pretty hard to refute a 99th generation listening test!

Test #2
I built a custom clock generator and put it in a DAT machine. On purpose, I increased the jitter of that clock generator to the point that a dubbing DAT machine almost could not lock to the signal from the jittery souce DAT. The sound coming out of the D/A converter of the dubbing DAT was entirely distorted, completely unlistenable. However, when played back, the dub had no audible distortion at all!

These are two scientifically-created proofs of an already well-understood digital "axiom," that the process of loading and storing digital data onto a storage medium effectively (or virtually) cancels the audible jitter coming in.

Does copying to hard disk deteriorate the sound of the source?
If you copy from a jittery source to a hard disk-recorder and later create a CDR from that hard disk, will this result in a jittery CDR? I cannot reach this conclusion based on personal listening experience. In most cases, the final CDR sounds better than the source, as auditioned direct off the hard disk! I must admit it is frustrating to listen to "degraded" sources and not really know how it is going to sound until you play back the final CDR.

Please note that I perform all my listening tests at Digital Domain through the same D/A converter, and that converter is preceded by an extremely powerful jitter-reduction device. Surprisingly, I can still hear some variation in source quality, depending on whether I am listening to hard disk, CDR, 20-bit tape, or DAT. The ear is an incredibly powerful "jitter detector"!

Quiz
Is it all right to make a digital chain of two or more DAT machines in record? The answer: During record you may hear a subtle loss of resolution due to increased jitter. However, the cumulative jitter in the chain will be reduced on playback. But we advise against chaining machines; it is safer to use a distribution amplifier (like the FCN-1) to feed multiple machines, because if one machine or a cable fails, the failure will not be passed on to another machine in line.

Can Compact Discs contain jitter?
When I started in this business, I was skeptical that there could be sonic differences between CDs that demonstrably contained the same data. But over time, I have learned to hear the subtle (but important) sonic differences between jittery (and less jittery) CDs. What started me on this quest was that CD pressings often sounded deteriorated (soundstage width, depth, resolution, purity of tone, other symptoms) compared to the CDR master from which they were made. Clients were coming to me, musicians with systems ranging from $1000 to $50,000, complaining about sonic differences that by traditional scientific theory should not exist. But the closer you look at the phenomenon of jitter, the more you realize that even minute amounts of jitter are audible, even through the FIFO (First in, First Out) buffer built into every CD player.

CDRs recorded on different types of machines sound different to my ears. An AES-EBU (stand-alone) CD recorder produces inferior-sounding CDs compared to a SCSI-based (computer) CD recorder. This is understandable when you realize that a SCSI-based recorder uses a crystal oscillator master clock. Whenever its buffer gets low, this type of recorder requests data on the SCSI buss from the source computer and thus is not dependent on the stability of the computer's clock. In contrast, a stand-alone CD recorder works exactly like a DAT machine; it slaves its master clock to the jittery incoming clock imbedded in the AES/EBU signal. No matter how effective the recorder's PLL at removing incoming jitter, it can never be as effective as a well-designed crystal clock.

I've also observed that a 4X-speed SCSI-based CDR copy sounds inferior to a double-speed copy and yet again inferior to a 1X speed copy.

Does a CD copy made from a jittery source sound inferior to one made from a clean source? I don't think so; I think the quality of the copy is solely dependent on clocking and mechanics involved during the transfer. Further research should be done on this question.

David Smith (of Sony Music) was the first to point out to me that power supply design is very important to jitter in a CD player, a CD recorder, or a glass mastering machine. Although the FIFO is supposed to eliminate all the jitter coming in, it doesn't seem to be doing an adequate job. One theory put forth by David is that the crystal oscillator at the output of the FIFO is powered by the same power supply that powers the input of the FIFO. Thus, the variations in loading at the input to the FIFO are microcosmically transmitted to the output of the FIFO through the power supply. Considering the minute amounts of jitter that are detectable by the ear, it is very difficult to design a power supply/grounding system that effectively blocks jitter from critical components. Crystal oscillators and phase locked loops should be powered from independent supplies, perhaps even battery supplies. A lot of research is left to be done; one of the difficulties is finding measurement instruments capable of quantifying very low amounts of jitter. Until we are able to correlate jitter measurements against audibility, the ear remains the final judge. Yet another obstacle to good "anti-jitter" engineering design is engineers who don't (or won't) listen. The proof is there before your ears!

David Smith also discovered that inserting a reclocking device during glass mastering definitely improves the sound of the CD pressing. Correlary question: If you use a good reclocking device on the final transfer to Glass Master, does this cancel out any jitter of previous source or source(s) that were used in the pre-production of the premaster? Answer: We're not sure yet!

Listening tests
I have participated in a number of blind (and double-blind) listening tests that clearly indicate that a CD which is pressed from a "jittery" source sounds worse than one made from a less jittery source. In one test, a CD plant pressed a number of test CDs, simply marked "A" or "B". No one outside of the plant knew which was "A" and which "B." All listeners preferred the pressing marked "A," as closer to the master, and sonically superior to "B." Not to prolong the suspense, disc "A" was glass mastered from PCM-1630, disc "B" from a CDR.

Attention CD Plants--a New Solution to the Jitter Problem from Sony
In response to pressure from its musical clients, and recognizing that jitter really is a problem, Sony Corporation has decided to improve on the quality of glass mastering. The result is a new system called (appropriately) The Ultimate Cutter. The system can be retrofitted to any CD plant's Glass mastering system for approximately $100,000. The Ultimate Cutter contains 2 gigabytes of flash RAM, and a very stable clock. It is designed to eliminate the multiple interfering clocks and mechanical irregularities of traditional systems using 1630, Exabyte, or CD ROM sources. First the data is transferred to the cutter's RAM from the CD Master; then all interfering sources may be shut down, and a glass master cut with the stable clock directly from RAM. This system is currently under test, and I look forward to hearing the sonic results.

Can Jitter in a Chain be Erased or Reduced?
The answer, thankfully, is "yes.". Several of the advanced D to A converters now available to consumers contain jitter reduction circuits. Some of them use a frequency-controlled crystal oscillator to average the moment to moment variations in the source. In essence, the clock driving the D/A becomes a stable crystal, immune to the pico- or nano-second time-base variations of jittery sources. This is especially important to professionals, who have to evaluate the digital audio during recording, perhaps at the end of a chain of several Phase Locked Loops. Someday all D to A converters will incorporate very effective jitter-reduction circuits.

Good Jitter vs. Bad Jitter
The amount of jitter is defined by how far the time is drifting. Original estimates of acceptable jitter in A/D and D/A converters were around 100 to 200 picoseconds (pS). However, research into oversampling converters revealed that jitter below 10 pS is highly desirable. For D/A converters, the amount of jitter is actually less important than the type of jitter, for some types of jitter are audibly more benign than others (I repeat: jitter does not affect D-D dubs, it only affects the D to A converter in the listening chain).

There are three different "types" of jitter:

1. The variations in the time base which are defined as jitter are regular and periodic (possibly sinusoidal)
2. The variations are random (incoherent, white noise)
3. The variations are related to the digital audio signal

Jitter can also be a combination of the above three.

Periodic fluctuations in the time base (#1 above) can cause spurious tones to appear at low levels, blocking our ability to hear critical ambient decay and thus truncating the dynamic range of the reproduction. Often this type of jitter is caused by clock leakage. It is analogous to scrape flutter in analog recorders.

On the other hand, Gaussian, or random jitter (#2 above, usually caused by a well-behaved Phase Locked Loop wandering randomly around the nominal clock frequency) is the least audible type. In addition to adding some additional noise at high frequencies, gaussian jitter adds a small perfume of hiss at the lowest levels, which may or may not be audible, and may or may not mask low level musical material. Sometimes, this type of jitter puts a "veil" on the sound. This veiling is not permanent (unlike the effects of dither, which are generally permanent), and will go away with a proper reclocking circuit into the D/A converter.

Finally, timing variations related to the digital audio signal (#3 above) add a kind of intermodulation distortion that can sound quite ugly.

More to Come
Jitter bibliography and credits. Clarifications of some apparent contradictions in the above essay.

While you're waiting for "The Jitter Bible," I urge you to listen, listen, listen, and see if you hear the problems of jitter in your audio systems, where and when they seem to occur.

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Part I: The 20th Century
Dealing With Peaks

Overs, levels, and headroom, how to get the most from your equipment

Digital recording is simple--all you do is peak to 0 dB and never go over! And things remain that simple until you discover one plugin or processor telling you a signal peaks to -1 dB while another meter (e.g. in your DAW) shows an OVER level, yet your digital outboard processor tells you it just reaches 0 dB! This article will explore concepts of the digital OVER, machine meters, loudness, and take a fresh look at the common practices of dubbing and level calibration.

 

Section I: Digital Meters and OVER Indicators
Manufacturers often have to pack a lot in their product, therefore compromising on meter design and accuracy to cut costs. A few outboard machines' meters are driven from analog circuitry, a definite source of inaccuracy. Even manufacturers who drive their meters digitally (by the values of the sample numbers) cut costs by putting large gaps on the meter scale (avoiding costly illuminated segments), using inaccurate calculations and/or time constants or by just not translating the values right to the visible meter. As a result, there may be a -3 point and a 0 dB point, with a big no man's land in between and the values not being represantative for the signals momentary peak-level. The manufacturer may feel he's doing you a favor by making the meter read 0 even if the actual level is between -1 and 0, or by setting the threshhold of the OVER indicator inaccurately or too conservatively (longbefore an OVER actually occurs). Even if the meter has a segment at every decibel, on playback, the plugin or DAW may not be able to tell the difference between a level of 0 dBFS (FS = Full Scale) and an OVER. I would question the machine's manufacturer if the OVER indicator lights on playback; it's probably a simple 0 dB detector rather than an OVER indicator.

There's only one way around this problem. Get a calibrated digital meter. Every studio should have one or two. There are lots of choices, from Dorrough, DK, Mytek, NTT, Pinguin, Sony, and others, each with unique features (including custom decay times and meter scales), but all the good meters agree on one thing: the definition of the highest measured digital audio level. A true digital audio meter reads the numeric code of the digital audio, and converts that to an accurate reading. A good digital audio meter can also distinguish between 0 dBFS and an OVER.

The Paradox of the Digital OVER
If digital levels cannot exceed 0 dB (by definition, there's nothing higher), then how can a digital signal go OVER? One way a signal can go OVER is during recording from an analog source. Of course the digitally encoded level cannot exceed 0 dBFS, but a level sensor in an A/D converter causes the OVER indicator to illuminate if the analog level is greater than the voltage equivalent to 0 dBFS. If the recordist does not reduce the analog record level, then a maximum level of 0 dB will be recorded for the duration of the overload, producing a nicely distorted square wave. There is a simple (digital) way of detecting if an OVER had occurred, even on playback--by looking for consecutive samples at 0 dB, which is a square wave. A specialized digital meter determines an OVER by counting thenumber of samples in a row at 0 dB. The Sony 1630 OVER standard is three samples, because it's fair to assume that the analog audio level must have exceeded 0 dB somewhere between sample number one and three. Three samples is a very conservative standard--most authorities consider distortion lasting only 33 microseconds (three samples at 44.1 KHz) to be inaudible. Manufacturers of digital meters often provide a choice of setting the OVER threshold to 4, 5, or 6 contiguous samples, but in this case it's better to be conservative. Even 6 samples is hard to hear on many types of music, so if you stick with the 3-sample standard, you'll guarantee that virtually all audible OVERs will be nipped in the bud, or at least detected! Once you've used a good digital meter, you'll never want to go back to the built-in kind.
In the diagram below, a positive-going analog signal goes OVER in the area above the dotted line.

Using External A/D Converters or Processors
There is no standard for communicating OVERs on an AES/EBU or S/PDIF line. So if you're using an external A/D converter and feed the signal into any machine, the OVER indicator there will probably not function properly or at all. I advise ignoring the indicator if it does light up, unless the manufacturer confirms that it's a sample counting OVER indicator. They'll probably reveal that it's an analog-driven level detector. Some external A/D converters do not have OVER indicators, so in this case, there's no substitute for an accurate external meter; without one I would advise not exceeding -1 dB on the feeded machine.

When making a digital dub through a digital processor you'll find most do not have accurate metering (be sure to read The Secrets of Dither before using any digital processor). Equalizer or processor sections can cause OVERs. Contrary to popular belief, an OVER can be generated even if a filter is set for attenuation instead of boost, because filters can ring. Digital processors can also overload internally in a fashion undetectable by a digital meter. Cascaded internal stages may "wrap around" when they overload, without transferring OVERs to the output. In those cases, a digital meter is not a foolproof OVER detector, and there's no substitute for the ear, but a good digital meter will catch most other transgressions. When you hear or detect an overload from a digital processor, try using the processor's digital input attenuator.

Practice Safe Levels
When recording to digital from an analog source, if you have an external digital meter set to 3 samples, then trust its OVER indicator and reduce gain slightly if it illuminates during recording. If you've been watching your levels prior to generating the OVER, chances are it will be an inaudible 3 sample OVER. You won't lose any meaningful signal-to-noise ratio, and you'll end up with a cleaner recording, especially when sending it for mastering. At the mastering studio, a file which is too hot can cause a digital EQ or sample rate converter to overload. There are ways around that, but not without complicating the mastering engineer's life.

Section II: How Loud is It?
Contrary to popular belief, the levels on a digital peak meter have (almost) nothing to do with loudness. For example, you're doing a direct to two-track recording (some engineers still work that way!) and you've found the perfect mix. Now, keep your hands off the faders, watch the levels to make sure they don't overload, and let the musicians make a perfect take. During take one, the performance reached -4 dB on the meter; and in take two, it reached 0 dB for a brief moment during a snare drum hit. Does that mean that take two is louder? If you answered "both takes are about the same loudness", you're probably right, because in general, the ear responds to average levels, not peak levels when judging loudness. If you raise the master gain of take one by 4 dB so that it, too reaches 0 dBFS, it will now sound 4 dB louder than take two, even though they both now measure the same on the peak meter.

Do not confuse the peak-reading meters on digital recorders with VU meters. Besides having a different scale, a VU meter has a much slower attack time than a digital peak meter. In PART II, we will discuss loudness in more detail, but let's summarize by saying that the VU meter responds more closely to the response of the ear. For loudness judgment, if all you have is a peak meter, use your ears. If you have a VU, use it as a guide, not an absolute, because the meter can be fooled (see PART II).

Did you know that an analog and digital recording of the same source sound very different in terms of loudness? Make an analog recording and a digital recording of the same music. Dub the analog recording to  the digital domain, peaking at 0 dB. The analog dub will sound about 6 dB louder than the all-digital recording! That's a lot. This is because the typical peak-to-average ratio of an analog recording is about 14 dB, compared with as much as 20 dB for an uncompressed digital recording. Analog tape's built-in compressor is a means of getting recordings to sound louder (oops, did I just reveal a secret?). That's why pop producers who record digitally may have to compress or limit to compete with the loudness of their analog counterparts.

The Myth of "Normalization"
Digital audio editing programs have a feature called "Normalization," a semi-automatic method of adjusting levels. The engineer selects all the segments (songs), and the computer grinds away, searching for the highest peak on the album. Then the computer adjusts the level of all the material until the highest peak reaches 0 dBFS. This is not a serious problem esthetically, as long as all the songs have been raised or lowered by the same amount. But it is also possible to select each song and "normalize" it individually. Since the ear responds to average levels, and normalization measures peak levels, the result can totally distort musical values. A compressed ballad will end up louder than a rock piece! In short, normalization should not be used to regulate song levels in an album. There's no substitute for the human ear.

Judging Loudness the Right Way
Since the ear is the only judge of loudness, is there any objective way to get a handle on how loud your CD will sound? The first key is to use a single D/A converter to reproduce all your digital sources. That way you can compare your CD in the making against other CDs, in the digital domain. Judge plugins, CDs, workstations, and digital processors through this single converter. Another important tool is a calibrated monitor level control with 1 dB per step settings. In a consistent monitoring environment, you can become familiar with the level settings of the monitor control for many genres of music, and immediately know how far you are (in dB) from your nearest competitor, just by looking at the setting of the monitor knob. At Digital Domain, we log all monitor settings used on a given project, so we can return to the same setting for revisions. In PART II, we will discuss how to use our knowledge to make a better system in the 21st Century.

The Moving Average Goes Up and Up...
Some of the latest digital processors permit making louder-sounding recordings than ever before. Today's mastering tools could make a nuclear bomb out of yesterday's firecrackers. But the sound becomes squashed, distorted and usually uninteresting. Visit my article on Compression for a more detailed description of the loudness race. While it seems the macho thing to do, you don't have to make your CD louder than the loudest current CD; try to make it sound better, which is much harder to do.

Section III: Calibrating Studio Levels
That concludes our production discussion. This next section is intended primarily for the maintenance engineer. Let's talk about alignment of studio audio levels. Stick around for a fresh perspective on level setting in the hybrid analog-digital studio.

Marking Tapes
dBm and dBv do not travel from house to house. These are measurements of voltages expressed in decibels. I once received a 1/4" tape in the mail marked "the level is +4 dBm." +4 dBm is a voltage (it's 1.23 volts, although the "m" stands for milliwatts). The 1/4" tape has no voltage on it, it doesn't have any idea whether it was made with a semi-pro level of 0 VU = -10 dBv or a professional level of +4. Voltages don't travel from house to house, only nanowebers per meter on analog tapes, and dBFS on digital tapes.
That doesn't diminish the importance of the analog reference level you use in-house. It's just irrelevant to the recipient of the tape. Just indicate the magnetic flux level which was used to coordinate with 0 VU. For example, 0 VU=400 nW/m at 1 KHz. Most alignment tapes have tables of common flux levels, where you'll find that 400 nW/M is 6 dB over 200 nW/m. Engineers often abbreviate this on the tape box as +6dB/200.

Deciding On an In-House Analog (voltage) Level
Just use the level provided by your console manufacturer, right? Well, maybe not. +4 dBv (reference .775 volts) may be a bad choice of reference level. Let's examine some factors you may not have considered when deciding on an in-house standard analog (voltage) level. When was the last time you checked the clipping point of your console and outboard gear? Before the advent of inexpensive 8-buss consoles, most professional consoles' clipping points were +24 dBv or higher. A frequent compromise in low-priced console design is to use internal circuits that clip around +20 dBv (7.75 volts). This can be a big impediment to clean audio, especially when cascading stages (how many of those amplifiers are between your source and your multitrack?). In my opinion, to avoid the "solid state edginess" that plagues a lot of modern equipment, the minimum clip level of every amplifier in your system should be 6 dB above the potential peak level of the music. The reason: Many opamps and other solid state circuits exhibit an extreme distortion increase long before they reach the actual clipping point. This means at least +30 dBv (24.5 volts RMS) if 0 VU is+4 dBv.

How Much Headroom is Enough?
Have you noticed that solid-state equipment starts to sound pretty nasty when used near its clip point? All other things being equal, the amplifier with the higher clipping point sounds better, in my opinion. Perhaps that's why tube equipment (with their 300 volt B+ supplies and headroom 30 dB or greater) often has a "good" name and solid state equipment with inadequate power supplies or headroom has a bad name.

Traditionally, the difference between average level and clip point has been called the headroom, but in order to emphasize the need for even more than the traditional amount of headroom, I'll call the space between the peak level of the music and the amplifier clip point a cushion. In the days of analog tape, a 0 VU reference of +4 dBv with a clipping point of +20 dBv provided reasonable amplifier headroom, because musical peak-to-average ratios were reduced to the compression point of the tape, which maxes out at around 14 dB over 0 VU. Instead of clipping, analog tape's gradual saturation curve produces 3rd and 2nd harmonics, much gentler on the ear than the higher order distortions of solid state amplifier clipping.

But it's a different story today, where the peak-to-average ratio of raw, unprocessed digital audio tracks can be 20 dB. Adding 20 dB to a reference of +4 dBv results in +24 dBv, which is beyond the clipping point of many so-called professional pieces of gear, and doesn't leave any room for a cushion . If you adapt an active balanced output to an unbalanced input, the clipping point reduces by 6 dB, so the situation becomes proportionally worse (all those headroom specs have to be reduced by 6 dB if you unbalance an amplifier's output). Be particularly suspicious of consoles that are designed to work at either professional or semi-pro levels. To meet price goals, manufacturers often compromise on headroom in professional mode, making the so-called semi-pro mode sound cleaner! You'll be unpleasantly surprised to discover that many consoles clip at +20 dBv, meaning they should never be using a professional reference level of +4 dBv (headroom of only 16 dB and no cushion). Even if the console clips at +30 dBv (the minimum clipping point I recommend), that only leaves a 6 dB cushion when reproducing music with 20 dB peak-to-average ratio. That's why more and more high-end professional equipment have clipping points as high as +37 dBv (55 volts!). To obtain that specification, an amplifier must use very high output devices and high-voltage power supplies. Translation--better sound.

To summarize, make sure the clip point of all your analog amplifiers is at least 6 dB (preferably 12 or more dB) above the peak level of analog material that will run in the system. I call this additional headroom the cushion.

How can you increase the cushion in your system, short of junking all your distribution amplifiers and consoles for new ones? One way to solve the problem is to recalibrate all your VU meters. You will not lose significant signal-to-noise ratio if you set 0 VU= 0 dBv or even -4 dBv (not an international standard, but a decent compromise if you don't want to throw out your equipment, and you have the expertise to make this standard stick throughout your studio). Try it and let me know if things sound cleaner in your studio.

Once you've decided on a standard analog reference level, calibrate all your analog-driven VU meters to this level. Here's a diagram describing the concept of cushion.

Dubbing and Copying - Translating between analog and digital points in the system
Let's discuss the interfacing of analog devices equipped with VU meters and digital devices equipped with digital (peak) meters. When you calibrate a system with sine wave tone, what translation level should you use? There are several de facto standards. Common choices have been -20 dBFS, -18 dBFS, and -14 dBFS translating to 0 VU. I'd like to see accurate calibration marks in digital recorders and DAWs at -12, -14, -18, and -20 dB, which covers most bases. Most of the external digital meters provide means to accurately calibrate at any of these levels.

How do you decide which standard to use? Is it possible to have only one standard? What are the compromises of each?

To make an educated decision, ask yourself: What is my system philosophy?

• Am I interested in maintaining headroom and avoiding peak clipping or do I want the highest possible signal-to-noise ratio at all times?
• Do I need to simplify dubbing practices or am I willing to require constant supervision during dubbing (operator checks levels before each dub, finds the peaks, and so on)?
• Am I adjusting levels or processing dynamics--mastering for loudness and consistency with only secondary regard for the peak level?

Consider your typical musical sources. Are your sources totally digital (DDD)? Did they pass through extreme processing (compression) or through analog tape stages? Pure, unprocessed digital sources, particularly individual tracks on a multitrack, will have peak levels 18 to 20 dB above 0 VU. Whereas processed mixdowns will have peak-to-average ratios of up to 18 dB (rarely up to 20). Analog tapes will have peak levels up to 14 dB, almost never greater. And that's how the three most common choices of translation numbers (-18,-20, and -14) were derived.

Broadcast Studios
In Broadcast, Practicality is our object, simplifying day-to-day operation, especially if your consoles are equipped with VU meters and your recorders are digital. In broadcast studios, it is desirable to use fixed, calibrated input and output gains on all equipment. My personal recommendation for the vast majority of studios is to standardize on reference levels of -20 dBFS ~0 VU, particularly when mixing to 2-track digital from live sources or tracking live to multitrack digital. If you're watching the console's VU meters, you will probably never clip a digital tape if you use -20 dBFS as a reference.
For a busy recording studio that does most of its mixing, recording and dubbing to harddisc, standardizing on -20 dBFS will simplify the process. Recording studios who decide on -18 dBFS ~0 VU will run into occasional digital clipping. That's why I'm against -18 dBFS as a standard for recording studios using VU meters for recording.

If you standardize on a -20 dBFS reference, the more compressed your musical material, the more signal-to-noise ratio you seem to be throwing away, but this is not true. If your source is analog tape, you might throw away 6 or more dB of signal, but this is less important than maintaining the convenience of never having to adjust dubbing levels on equipment. Furthermore, the ear judges noise level by average levels, and if the crest factor of your material is 6 dB less, it will seem just as loud as the uncompressed material peaking to 0 dBFS, you will not have to turn up your monitor, and you will not hear additional noise. Remember: analog tapes typically sound 6 dB louder than digital tapes, if peaked to the same peak level.

A -20 reference is only a potential problem when dubbing from digital source to analog tape. In many cases, you can accept the innocuous 6 dB compression. We've been enjoying that for years when we mixed from live material on VU-equipped console direct to analog tape. When making dubs to analog for archival purposes, choose a tape with more headroom, or use a custom reference point (-14 to -18 dBFS), as the goal is to preserve transients for the enjoyment of future listeners. A calibrated peak level meter on the analog machine will tell you what it's doing more than a VU meter. For archival purposes, I prefer to use the headroom of the new high-output tapes for transient clarity, rather than to jack up the flux level for a better signal-to-hiss ratio.

If working in a broadcast facility which seems no live (uncompressed) material, then for the broadcast dubbing room, -14 is a good number (dubbing between analog and digital tapes). -18 is a safe all-around reference for all the other A/D/A converters in the broadcast complex, since most of the material will have 18 dB or lower peak-to average ratio, and occasional clipping maybe tolerated.

Mastering Studios
mastering studios are working more frequently in 20-bit or 24-bit. In Part II, I suggest the 21st Century approach to mastering.

Analog PPMs
Analog PPMs have a slower attack time than digital PPMs. When working with a digital recorder, a live source, and desk equipped with analog PPM, I suggest a 5 dB "lead." In other words, align the highest peak level on the analog PPM to -5 dBFS with sine wave tone.

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Part II: How To Make Better Recordings in the 21st Century - An Integrated Approach to Metering, Monitoring, and Leveling Practices.

(includes a description of the K-System, an integrated system of metering and monitoring)

Updated from the article published in the September 2000 issue of the AES Journal by Bob Katz

A: Two-Channel

For the last 30 years or so, film mix engineers have enjoyed the liberty and privilege of a controlled monitoring environment with a fixed (calibrated) monitor gain. The result has been a legacy of feature films, many with exciting dynamic range, consistent and natural-sounding dialogue, music and effects levels. In contrast, the broadcast and music recording disciplines have entered a runaway loudness race leading to chaos at the end of the 20th century. I propose an integrated system of metering and monitoring that will encourage more consistent leveling practices among the three disciplines. This system handles the issue of differing dynamic range requirements far more elegantly and ergonomically than in the past. We're on the threshold of the introduction of a new, high-resolution consumer audio format and we have a unique opportunity to implement a 21st Century approach to leveling, that integrates with the concept of Metadata. Let's try to make this a worldwide standard to leave a legacy of better recordings in the 21st Century.

History of the VU meter
On May 1, 1999, the VU meter celebrated its 60th birthday. 60 years old, but still widely misunderstood and misused. The VU meter has a carefully-specified time-dependent response to program material which this paper refers to as "Average," or "averaging", but means the particular VU meter response. This instrument was intended to help program producers create consistent loudness amongst program elements, but was not a suitable measure of when the recording medium was being exceeded, or overloaded. Therefore the meter's designers assumed that the recording medium would have at least 10 dB Headroom over 0 VU, like the analog media then in use.

Summary of VU Inconsistencies and Errors

In General, the meter's ballistics, scale, and frequency response all contribute to an inaccurate indicator. The meter approximates momentary loudness changes in program material, but reports that moment-to-moment level differences are greater than the ear actually perceives.

Ballistics
The meter's ballistics were designed to "look good" with spoken word. Its 300 ms integration time gives it a syllabic response, which looks very "comfortable" with speech, but doesn't make it accurate. One time constant cannot sum up the complex multiple time constants required to model the loudness perception of the human listener. Skilled users soon learned that an occasional short "burst" from 0 to +3 VU would probably not cause distortion, and usually was meaningless as far as a loudness change.

Scale
In 1939, logarithmic amplifiers were large and cumbersome to construct, and it was desirable to use a simple passive circuit. The result is a meter where every decibel of change is not given equal merit. The top 50% of the physical scale is devoted to only the top 6 dB of dynamic range, and the meter's useable dynamic range is only about 13 dB. Not realizing this fundamental fact, inexperienced and experienced operators alike tend to push audio levels and/or compress them to stay within this visible range. With uncompressed material, the needle fluctuates far greater than the perceived loudness change and it is difficult to distinguish compressed from uncompressed material by the meter. Soft material may hardly move the meter, but be well within the acceptable limits for the medium and the intended listening environment.

Frequency response
The meter's relatively flat frequency response results in extreme meter deflections that are far greater than the perceived loudness change, since the ear's response is non-linear with respect to frequency. For instance, when mastering reggae music, which has a very heavy bass content, the VU meter may bounce several dB in response to the bass rhythm, but perceived loudness change is probably less than a dB.

Lack of conformance to standards
There are large numbers of improperly-terminated mechanical VU meters and inexpensively-constructed indicators which are labelled "VU" in current use. These disparate meters contribute to disagreements among program producers reading different instruments. A true VU meter is a rather expensive device. It's not a VU meter unless it meets the standard.

Over the past 60 years, psychoacousticians have learned how to measure perceived loudness much better than a VU. Despite all these facts, the VU meter is a very primitive loudness meter. In addition, current digital technology permits us to easily correct the non-linear scale, its dynamic range, ballistics,and frequency response.

II. Current-day levelling problems

In the music and broadcast industries, chaos currently prevails. Here is a waveform taken from a digital audio workstation, showing three different styles of music recording. The time scale is about 10 minutes total, and the vertical scale is linear, +/- 1 at full digital level, 0.5 amplitude is 6 dB below full scale. The "density" of the waveform gives a rough approximation of the music's dynamic range and Crest Factor (headroom for peaks above the average level). On the left side is a piece of heavily compressed pseudo "elevator music" I constructed for a demonstration at the 107th AES Convention. In the middle is a four-minute popular compact disc single produced in 1999, with sales in the millions. On the right is a four-minute popular rock and roll recording made in 1990 that's quite dynamic-sounding for rock and roll of that period. The perceived loudness difference between the 1990 and 1999 CDs is greater than 6 dB, though both peak to full scale. Auditioning the 1999 CD, one mastering engineer remarked "this CD is a lightbulb! The music starts, all the meterlights come on, and it stays there the whole time." To say nothing about the distortion. Are we really in the business of making square waves?

The average level of popular music compact discs continues to rise. Popular CDs with this problem are becoming increasingly prevalent, coexisting with discs that have beautiful dynamic range and impact, but whose loudness (and distortion level) is far lower. There are many technical, sociological and economic reasons for this chaos that are beyond the scope of this paper. Let's concentrate on what we can do as an engineering body to help reduce this chaos, which is a disservice to the consumer. It's also an obstacle to creating quality program material in the 21st century. What good is a 24-bit/96 kHz digital audio system if the programs we create only have 1 bit dynamic range?

Is this what will happen to the next generation carrier? (e.g. DVD-A, SACD). It will, if we don't take steps to stop it. Unlike with the LP, there is no PHYSICAL limit to the average level we can place on a digital medium. Note that there is a point of diminishing returns above about -14 dBFS. Dynamic inversion begins to occur and the program material usually stops sounding louder because it loses clarity and transient response.

III. The Magic of "83" with Film Mixes
In the music world, everyone currently determines their own average record level, and adjusts their monitor accordingly. With no standard, subjective loudness varies from CD to CD in popular music as much as 10-12 dB, which is unacceptable by any professional standard. But in the film world, films are consistent from one to another, because the monitoring gain has been standardized. In 1983, as workshops chairman of the AES Convention, I invited Tomlinson Holman of Lucasfilm to demonstrate the sound techniques used in creating the Star Wars films. Dolby systems engineers labored for two days to calibrate the reproduction system in New York's flagship Ziegfeld theatre. Over 1000 convention attendees filled the theatre center section. At the end of the demonstration, Tom asked for a show of hands. "How many of you thought the sound was too loud?" About four hands were raised. "How many thought it was too soft?" No hands. "How many thought it was just right?" At least 996 audio engineers raised their hands.

This is an incredible testament to the effectiveness of the 83 dB SPL reference standard proposed by Dolby's Ioan Allen in the mid-70's, originally calibrated to a level of 0 VU for use with analog magnetic film. The choice of 83 dB SPL has stood the test of time, as it permits wide dynamic range recordings with little or no perceived system noise when recording to magnetic film or 20-bit digital. Dialogue, music and effects fall into a natural perspective with an excellent signal-to-noise ratio and headroom. A good film mix engineer can work without a meter and do it all by the monitor, using the meter simply as a guide. In fact, working with a fixed monitor gain is liberating, not limiting. When digital technology reached the large theatre, the SMPTE attached the SPL calibration to a point below full scale digital. When we converted to digital technology, the VU meter was rapidly replaced by the peak program meter.

When AC-3 and DTS became available for home theatre, many authorities recommended lowering the monitor gain by 6 dB because a typical home listening room does not accomodate high SPLs and wide dynamic range. If a DVD contains the wide range theatre mix, many home listeners complain that "this DVD is too loud", or "I lose the dialogue when I turn the volume down so that the effects don't blast." With reduced monitor gain, the soft passages become too soft. For such listeners, the dynamic range may have to be reduced by 6 dB (6 dB upward Compression, or dynamic range reduction) in order to use less monitor gain.
Metadata are coded data which contain information about signal dynamics and intended loudness; this will resolve the conflict between listeners who want the full theatrical experience and those who need to listen softly. But without metadata there are only two solutions: a) to compromise the audio soundtrack by compressing it, or better, b) use an optional compressor for the home system. With the later approach the source audio is uncompromised.

IV. The Magic of "-6 dB" Monitor Gain for the Home
In the 21st century, home theatre, music, and computers are becoming united. Many, if not most, consumers will eventually be auditioning music discs on the same system that plays broadcast television, home theatre (DVDs), and possibly even web-audio, e.g. MP3. Music-only discs are often used as casual or background music, but I am specifically referring to foreground music that the discerning consumer or audiophile will play at normal or full "enjoyment" loudness.

With the integration of media into a single system, it is in the direct interest of music producers to think holistically and unite with video and film producers for a more consistent consumer audio presentation. Music producers experimenting with 5.1 surround must pay more than casual attention to monitor level calibration. They have already discovered the annoyance that a typical pop CD will blast the sound system when inserted into a DVD player after a movie has been played. Recently a DVD and soundtrack CD were produced of the classic rock music movie Yellow Submarine. Reviewers complained that the CD is much louder and less dynamic than the DVD. Audio CDs should not be degraded for the sake of a "loudness competition". CDs can and should be produced to the same audio quality standard as the DVD.

New program producers with little experience in audio production are coming into the audio field from the computer, software and computer games arena. We are entering an era where the learning curve is high, engineer's experience is low, and the monitors they use to make program judgments are less than ideal. It is our responsibility to educate engineers on how to make loudness judgments. A plethora of peak-only meters on every computer, DAT machine and digital console do not provide information on program loudness. Engineers must learn that the sole purpose of the peak meter is to protect the medium and that something more like average level affects the program's loudness. Bear in mind that the bandwidth and frequency distribution of the signal also affect program loudness.

As a music mastering engineer, I have been studying the perceived loudness of music compact discs for over 15 years. Around 1993, I installed a 1 dB/per step monitor control for repeatability. In an effort to achieve greater consistency from disc to disc, I made it a point to try to set the monitor gain first, and then master the disc to work well at that monitor gain.

In 1996, we measured that monitor gain, and found it to be 6 dB less than the film-standard for most of the pop music we were mastering. To calibrate a monitor to the film-standard, play a standardized pink noise calibration signal whose amplitude is -20 dB FS RMS, on one channel (loudspeaker) at a time. Adjust the monitor gain to yield 83 dB SPL using a meter with C-weighted, slow response. Call this gain 0 dB, the reference, and you will find the pop-music "standard" monitor gain at 6 dB below this reference.
By now, we've mastered hundreds of pop CDs working at monitor gain 6 dB below the reference, with very satisfied clients. However, if monitor gain is further reduced, average recorded level tends go up because the mastering engineer seeks the same loudness to the ears. Since the average program level is now closer to the maximum permissible peak level, more compression/limiting must be used to keep the system from overloading. Increased compression/limiting is potentially damaging to the program material, resulting in a distorted, crowded, unnatural sound. Clients must be informed that they can't get something for nothing; a hotter record means lower sound quality.

Mastering and the Loudness Race
By 1997, some music clients were complaining that their reference CDs were "not hot enough", a tragic testimony on the loudness race which is slowly destroying the industry. Each client wants his CD to be as loud as or louder than the previous "winner", but every winner is really a loser. Fueling that race are powerful digital compressors and limiters which enable mastering engineers to produce CDs whose average level is almost the same as the peak level! There is no precedent for that in over 100 years of recording. We end up mastering to the lowest common denominator, and fight desperately to avoid that situation, wasting a lot of time showing clients that the sound quality suffers as the average level goes up. The psychoacoustic problem is that when two identical programs are presented at slightly differing loudness, the louder of the two often appears "better" in short term listening. This explains why CD loudness levels have been creeping up until sound quality is so bad that everyone can perceive it. Remember that the loudness "race" has always been an artificial one, since the consumer adjusts their volume control according to each record anyway.

In addition, it should be more widely known that hyper-compressed recordings do not play well on the radio. They sound softer and seriously distorted, pointing out that the loudness race has no winners, even in radio airplay. The best way to make a "radio-ready" recording is not to squash it, but rather produce it with the typical peak to average ratios that have worked for about a hundred years.

As the years went on, trying to "hold the fort", I gradually raised the average level of mastered CDs only when requested, which forced the monitor gain to be reduced from 1 to several dB. For every decibel of increased average level, considerably more damage is done to the sound. We often note severe processor distortion when the monitor gain falls below -6 dB. Consumers find their volume controls at the bottom of their travel, where a small control movement produces awkward level changes.

V. The relationship between SPL and 0 VU
In 1994, I installed a pair of Dorrough meters, in order to view the average and peak level simultaneously on the same scale. These meters use a scale with 0 "average" (a quasi-VU characteristic I'll call "AVG") placed at 14 dB below full digital scale, and full scale marked as +14 dB. music mastering engineers often use this scale, since a typical stereo 1/2" 30 IPS analog tape has approximately 14 dB headroom above 0 VU.

The next step is to examine a simple relationship between the 0 AVG level and the sound pressure level. For typical pop productions, our monitor gain has been adjusted to -6 dB (below the standard reference, which yields 77dB SPL with -20 dBFS pink noise).

Since -20 dBFS reads -6 AVG, then 6 dB higher, or 0 AVG must be 83 dB SPL. In other words, we're really running average SPLs similar to the original theatre standard. The only difference is that headroom is 14 dB above 83 instead of 20. Running a sound pressure level meter during the mastering session confirms that the ear likes 0 AVG to end up circa 83 dB (~86 dB with both loudspeakers operating) on forte passages, even in this compressed structure. If the monitor gain is further reduced by 2 dB the mastering engineer judges the loudness to be lower, and thus raises average recorded level--and the AVG meter goes up by 2 dB. It's a linear relationship. This leads us to the logical conclusion that we can produce programs with different amounts of dynamic range (and headroom) by designing a loudness meter with a sliding scale, where the moveable 0 point is always tied to the same calibrated monitor SPL. Regardless of the scale, production personnel would tend to place music near the 0 point on forte passages.

VI. The K-System Proposal
The proposed K-System is a metering and monitoring standard that integrates the best concepts of the past with current psychoacoustic knowledge in order to avoid the chaos of the last 20 years.

In the 20th Century we concentrated on the medium. In the 21st Century,we should concentrate on the message. We should avoid meters which have 0 dB at the top--this discourages operators from understanding where the message really is. Instead, we move to a metering system where 0 dB is a reference loudness, which also determines the monitor gain. In use, programs which exceed 0 dB give some indication of the amount of processing (compression) which must have been used. There are three different K-System meter scales, with 0 dB at either 20, 14, or 12 dB below full scale, for typical headroom and SNR requirements. The dual-characteristic meter has a bar representing the average level and a moving line or dot above the bar representing the most recent highest instantaneous (1 sample) peak level.

Several accepted methods of measuring loudness exist, of varying accuracy (e.g., ISO 532, LEQ, Fletcher-Harvey-Munson, Zwicker and others, some unpublished).The extendable K-system accepts all these and future methods, plus providing a "flat" version with RMS characteristic. Users can calibrate their system's electrical levels with pink noise, without requiring an external meter. RMS also makes a reasonably-effective program meter that many users will prefer to a VU meter.

The three K-System meter scales are named K-20, K-14, and K-12. I've also nicknamed them the papa, mama, and baby meters. The K-20 meter is intended for wide dynamic range material, e.g., large theatre mixes, "daring home theatre" mixes, audiophile music, classical (symphonic) music, "audiophile" pop music mixed in 5.1 surround, and so on. The K-14 meter is for the vast majority of moderately-compressed high-fidelity productions intended for home listening (e.g. some home theatre, pop, folk, and rock music). And the K-12 meter is for productions to be dedicated for broadcast.

Note that full scale digital is always at the top of each K-System meter. The 83 dB SPL point slides relative to the maximum peak level. Using the term K-(N) defines simultaneously the meter's 0 dB point and the monitoring gain.

The peak and average scales are calibrated as per AES-17, so that peak and average sections are referenced to the same decibel value with a sine wave signal. In other words, +20 dB RMS with sine wave reads the same as +20 dB peak, and this parity will be true only with a sine wave. Analog voltage level is not specified in the K-system, only SPL and digital values. There is no conflict with -18 dBFS analog reference points commonly used in Europe.

VII. Production Techniques with the K-System
To use the system, first choose one of the three meters based on the intended application. Wide dynamic range material probably requires K-20 and medium range material K-14. Then, calibrate the monitor gain where 0dB on the meter yields 83 dB SPL (per channel, C-Weighted, slow speed). 0dB always represents the same calibrated SPL on all three scales, unifying production practices worldwide. The K-system is not just a meter scale, it is an integrated system tied to monitoring gain.

A manual for a certain digital limiter reads: "For best results, start out with a threshold of -6 dB FS". This is like saying "always put a teaspoon of salt and pepper on your food before tasting it." This kind of bad advice does not encourage proper production practice. A gain reduction meter is not an indication of loudness. Proper metering and monitoring practice is the only solution.

If console and workstation designers standardize on the K-System it will make it easier for engineers to move programs from studio to studio. Sound quality will improve by uniting the steps of pre-production (recording and mixing), post-production (mastering) and metadata (authoring) with a common "level" language. By anchoring operations to a consistent monitor reference, operators will produce more consistent output, and everyone will recognize what the meter means.

If making an audiophile recording, then use K-20, if making "typical" pop or rock music, or audio for video, then probably choose K-14. K-12 should be reserved strictly for audio to be dedicated to broadcast; broadcast recording engineers may certainly choose K-14 if they feel it fits their program material. Pop engineers are encouraged to use K-20 when the music has useful dynamic range.

The two prime scales, K-20 and K-14, will create a cluster near two different monitor gain positions. People who listen to both classical and popular music are already used to moving their monitor gains about 6 dB (sometimes 8 to 12 dB with the hottest pop CDs). It will become a joy to find that only two monitor positions satisfy most production chores. With care, producers can reduce program differences even further by ignoring the meter for the most part, and working solely with the calibrated monitor.

Using the Meter's Red Zone. This 88-90 dB+ region is used in films for explosions and special effects. In music recording, naturally-recorded (uncompressed) large symphonic ensembles and big bands reach +3 to +4 dB on the average scale on the loudest (fortissimo) passages. Rock and electric pop music take advantage of this "loud zone", since climaxes, loud choruses and occasional peak moments sound incorrect if they only reach 0dB (forte) on any K-system meter. Composers have equated fortissimo to 88-90+ dB since the time of Beethoven. Use this range occasionally, otherwise it is musically incorrect (and ear-damaging). If engineers find themselves using the red zone all the time, then either the monitor gain is not properly calibrated, the music is extremely unusual (e.g. "heavy metal"), or the engineer needs more monitor gain to correlate with his or her personal sensitivities. Otherwise the recording will end up overcompressed, with squashed transients, and its loudness quotient out of line with K-System guidelines.

Equal Loudness Contours
Mastering engineers are more inclined to work with a constant monitor gain. But many music mixing engineers work at a much higher SPL, and also vary their monitor gain to check the mix at different SPLs. I recommend that mix engineers calibrate your monitor attenuators so you can always return to the recommended standard for the majority of the mix. Otherwise it is likely the mix will not translate to other venues, since the equal-loudness contours indicate a program will be bass-shy when reproduced at a lower (normal) level.

Tracking/Mixing/Mastering
The K-System will probably not be needed for multitracking--a simple peak meter is probably sufficient. For highest sound quality, use K-20 while mixing and save K-14 for the calibrated mastering suite. If mixing to analog tape, work at K-20, and realize that the peak levels off tape will not exceed about +14. K-20 doesn't prevent the mix engineer from using compressors during mixing, but the author hopes that engineers will return towards using compression as an esthetic device rather than a "loudness-maker."

Using K-20 during mix encourages a clean-sounding mix that's advantageous to the mastering engineer. At that point, the producer and mastering engineer should discuss whether the program should be converted to K-14, or remain at K-20. The K-System can become the lingua franca of interchange within the industry, avoiding the current problem where different mix engineers work on parts of an album to different standards of loudness and compression.

When the K-System is not available
Current-day analog mixing consoles equipped with VUs are far less of a problem than digital models with only peak meters. Calibrate the mixdown A/D gain to -20 dBFS at 0 VU, and mix normally with the analog console and VUs. However, mixing consoles should be retro fitted with calibrated monitor attenuators so the mix engineer can repeatably return to the same monitor setting.

Compression is a powerful esthetic tool. But with higher monitor gain, less compression is needed to make material sound good or "punchy." For pop music, many K-14 presentations sound better than K-20, with skillfully-applied dynamics processing by a mastering engineer working in a calibrated room. But clearly, the higher the K-number, the easier it is to make it sound "open" and clean. Use monitor systems with good headroom so that monitor compression does not contaminate the judgment of program transients.

Adapting large theatre material to home use may require a change of monitor gain and meter scale. Producers may choose to compress the original 6-channel theatre master, or better, remix the entire program from the multi-track stems (submixes). With care, most of the virtues and impact of the original production can be maintained in the home. Even audiophiles will find a well-mastered K-14 program to be enjoyable and dynamic. It is desirable to try to fit this reduced-range mix on the same DVD as the wide-range theatre mix.

Multichannel to Stereo Reductions
The current legacy of loud pop CDs creates a dilemma because DVD players can also play CDs. Producers should try to create the 5.1 mix of a project at K-20. If possible, the stereo version should also be mixed and mastered at K-20. While a K-20 CD will not be as loud as many current pop CDs, it may be more dynamic and enjoyable, and there will not be a serious loudness jump compared to K-20 DVDs in the same player. If the producer insists on a "louder" CD, try to make it no louder than K-14, in which case there will only be 6 dB loudness difference between the DVD and the audio CD. Tell the producer that the vast majority of great-sounding pop CDs have been made at K-14 and the CD will be consistent with the lot, even if it isn't as hot as the current hypercompressed "fashion." It's the hypercompressed CD that's out of line, not the K-14.

Full scale peaks and SNR
It is a common myth that audible signal-to-noise ratio will deteriorate if a recording does not reach full scale digital. On the contrary, the actual loudness of the program determines the program's perceived signal-to-noise ratio. The position of the listener's monitor level control determines the perceived loudness of the system noise. If two similar music programs reach 0 on the K-system's average meter, even if one peaks to full scale and the other does not, both programs will have similar perceived SNR. Especially with 20-24 bit converters, the mix does not have to reach full scale (peak). Use the averaging meter and your ears as you normally would, and with K-20, even if the peaks don't hit the top, the mixdown is still considered normal and ready for mastering, with no audible loss of SNR.

Multipurpose Control Rooms
With the K-System, multipurpose production facilities will be able to work with wide-dynamic range productions (music,videos, films) one day, and mix pop music the next. A simultaneous meter scale and monitor gain change accomplishes the job. It seems intuitive to automatically change the meter scale with the monitor gain, but this makes it difficult to illustrate to engineers that K-14 really is louder than K-20.

A simple 1 dB per step monitor attenuator can be constructed, and the operator must shift the meter scale manually.

Calibrate the gain of the reproduction system power amplifiers or preamplifiers with the K-20 meter, and monitor control at the "83" or 0 dB mark. Operators should be trained to change the monitor gain according to the K-System meter in use.

Here is the K-20/RMS meter in close detail, with the calibration points.

Individuals who decide to use a different monitor gain should log it on the tape (file) box, and try to use this point consistently. Even with slight deviations from the recommended K(N) practice, the music world will be far more consistent than the current chaos. Everyone should know the monitor gain they like to use.

At left is a picture of an actual K-14/RMS Meter in operation at the Digital Domain studio, as implemented by Metric Halo labs in the program Spectrafoo for the Macintosh. Spectrafoo versions 3f17 and above include full K-System support and a calibrated RMS pink noise generator. Other meters that conform exactly with K-System guidelines have been implemented by Pinguin for PC, RME in their Digichek software, and Roger Nichols Digital (formerly Elemental audio) Inspector XL. The Dorrough and DK meters nearly meet K-System guidelines but an external RMS meter must be used for pink noise calibration since they use a different type of averaging. In practice with program material, the difference between RMS and other averaging methods is insignificant, especially when you consider that neither method is close enough to a true loudness meter. As of this date, 12/05/07, we are still awaiting a company that will implement the K-System with a loudness characteristic, such as Zwicker.

Audio Cassette Duplication
Cassette duplication has been practiced more as an art than a science, but it should be possible to do better. The K-System may finally put us all on the same page (just in time for obsolescence of the cassette format). It's been difficult for mastering engineers to communicate with audio cassette duplicators, finding a reference level we all can understand. A knowledgeable duplicator once explained that the tape most commonly used cannot tolerate average levels greater than +3 over 185 nW/m (especially at low frequencies) and high frequency peaks greater than about +5-6 are bound to be distorted and/or attenuated. Displaying crest factor makes it easy to identify potential problems; also an engineer can apply cassette high-frequency preemphasis to the meter. Armed with that information, an engineer can make a good cassette master by using a "predistortion" filter with gentle high-frequency compression and equalization. Meter with K-14 or K-20, and put test tone at the K-System reference 0 on the digital master. Peaks must not reach full scale or the cassette will distort. Apparent loudness will be less than the K-standard, but this is a special case.

Classical music
It's hard to get out of the habit of peaking our recordings to the highest permissible level, even though 24-bit systems have a theoretically 48 dB better signal-to-dither-ratio than 16-bit. It is much better for the consumer to have a consistent monitor gain than to peak every recording to full scale digital. I believe that attentive listeners prefer auditioning at or near the natural sound pressure of the original classical ensemble (see Footnote). The dilemma is that string quartets and Renaissance music, among other forms, have low crest factors as well as low natural loudness. Consequently, the string quartet will sound (unnaturally) much louder than the symphony if both are peaked to full scale digital.

I recommend that classical engineers mix by the calibrated monitor, and use the average section of the K-meter only as a guide. It's best to fix the monitor gain at 83 dB and always use the K-20 meter even if the peak level does not reach full scale. There will be less monitoring chaos and more satisfied listeners. However, some classical producers are concerned about loss of resolution in the 16-bit medium and may wish to peak all recordings to full scale. I hope you will reconsider this thought with 24 bit media or SACD.

Narrow Dynamic Range Pop Music
We can avoid a new loudness race and consequent quality reduction if we unite behind the K-System before we start fresh with high-resolution audio media such as DVD-A and SACD. Similar to the above classical music example, pop music with a crest factor much less than 14 dB should not be mastered to peak to full scale, as it will sound too loud.
Recommended:

1: Author with metadata to benefit consumers using equipment that supports metadata
2: If possible, master such discs at K-14
3: Legacy music, remasters from often overcompressed CD material should be reexamined for its loudness character. If possible, reduce the gain during remastering so the average level falls within K-14 guidelines. Even better, remaster the music from unprocessed mixes to undo some of the unnecessary damage incurred during the years of chaos. Some mastering engineers already have made archives without severe processing.

VIII. An Extendable System
Since the K-System is extendable to future methods of measuring loudness, program producers should mark their tape boxes or digital files with an indication which K-meter and monitor calibration was used. For example, "K-14/RMS," or "K-20/Zwicker." I hope that these labels will someday become as common as listings of nanowebers per meter and test tones for analog tapes. If a non-standard monitor gain was used, note that fact on the tape box to aid in post-production authoring and insertion of metadata.

IX. Metadata and the K-System
Dolby AC-3, MPEG2, AAC, and hopefully MLP will take advantage of metadata control words. Pre-production with the K-System will speed the authoring of metadata for broadcast and digital media. Music producers must familiarize themselves with how metadata affects the listening experience. First we'll summarize how the control word Dialnorm is used in digital television. Then we will examine how to take advantage of Dialnorm and MixLevel for music-only productions.

Dialnorm
Dialogue normalization, is used in digital television and radio as "ecumenical gain-riding". Program level is controlled at the decoder, producing a consistent average loudness from program to program; with the amount of attenuation individually calculated for each program. The receiver decodes the dialnorm control word and attenuates the level by the calculated amount, resulting in the "table radio in the kitchen" effect. In an unnatural manner, average levels of sports broadcasts, rock and roll, newscasts, commercials, quiet dramas, soap operas, and classical music all end up at the loudness of average spoken dialogue.

With Dialnorm, the average loudness of all material is reduced to a value of -31 dB FS (LEQ-A). Theatrical films with dialogue at around -27 dB FS will be reduced 4 dB. -31 corresponds not with musical forte, but rather mezzo-piano. For example, a piece of rock and roll, normally meant to be reproduced forte, may be reduced 10 or more dB, while a string quartet may only be reduced 4-5 dB at the decoder. The dialnorm value for a symphony should probably be determined during the second or third movement, or the results will be seriously skewed. We do want the forte passages to be louder than the spoken word! Rock and roll, with its more limited dynamic range, will be attenuated farther from "real life" than the symphony. However, unlike the analog approach, the listener can turn up his receiver gain and experience the original program loudness--without the noise modulation and squashing of current analog broadcast techniques. Or, the listener can choose to turn off dialnorm (on some receivers) and experience a large loudness variance from program to program.

Each program is transmitted with its full intended dynamic range, without any of the compression used in analog broadcasting--the listener will hear the full range of the studio mix. For example, in variety shows, the music group will sound pleasingly louder than the presenter. Crowd noises in sports broadcasts will be excitingly loud, and the announcer's mike will no longer "step on" the effects, because the bus compressor will be banished from the broadcast chain.

Mixlev
Dialnorm does not reproduce the dyamic range of real life from program to program. This is where the optional control word mixlev (mix level) enters the picture. The dialnorm control word is designed for casual listeners, and mixlev for audiophiles or producers. Very simply, mixlev sets the listener's monitor gain to reproduce the SPL used by the original music producer. Only certain critical listeners will be interested in mixlev. If the K-system was used to produce the program, then K-14 material will require a 6 dB reduction in monitor gain compared to K-20, and so on. Mixlev will permit this change to happen automatically and unattended. Attentive listeners using mixlev will no longer have to turn down monitor gains for string quartets, or up for the symphony or (some) rock and roll.

The use of dialnorm and mixlev can be extended to other encoded media, such as DVD-A. Proper application of dialnorm and mixlev, in conjunction with the K-System for pre-production practice--will result in a far more enjoyable and musical experience than we currently have at the end of the 20th century of audio.

X. In Conclusion
Let's bring audio into the 21st century. The K-system is the first integrated approach to monitoring, levelling practices, metering and metadata.

B: Multichannel
There's good news for audio quality: 5.1 surround sound. Current mixes of popular music that I have listened to in 5.1 sound open, clear, beautiful, yet also impacting. I've done meter measurements and listening to a few excellent 20 and 24 bit 5.1 mixes, and they all fall perfectly into the K-20 Standard. Monitor gain ran from 0 dB to -3 dB, mostly depending on taste, as it was perfectly comfortable to listen to all of these particular recordings at 0 dB (reference RP 200).

What became clear while watching the K-20 meter is that the best engineers are using the peak capability of the 5.1 system strictly for headroom. It is possible that I didn't see a single peak to full scale (+20 on the K-20 Meter) on any of these mixes. The averaging portion of the meter operated just as in my recommendations, with occasional peaks to +4 on some of the channels.

Monitor calibration made on an individual speaker basis worked extremely well, with the headroom in each individual channel tending to go up as the number of channels increases. This is simply not a problem with 24 bit (or even 20 bit) recording. System hiss is not evident at RP 200 monitor gains with long-wordlength recording, good D/A converters, modern preamps and power amplifiers.

Another question is: Should we have an overall meter calibrated to a total SPL? If so, what should that SPL be? My initial reactions are that an overall meter is not necessary, at least in mix situations where mix engineers use calibrated monitoring and monitors with good headroom.

Another positive thought. I've been giving 5.1 seminars sponsored by TC, Dynaudio, and DK Meters. To begin the show, I played two stereo masters that I had mastered, and demonstrated some very sophisticated techniques to bump them up (transparently) to 5.1. This is a growing field, and you'll see increasing techniques for doing this, especially when the record company wants a DVD or DVD-A remaster without (horrors) having to pay for a remix.

The good news is I found that the true 5.1 mixes by George Massenburg and others that I was demonstrating sounded so OPEN and clear and beautiful that even I was embarrassed to start from a 24-bit version of my own two masters. I had to remaster the two pieces with about 2 to 4 dB LESS LIMITING in order to make them COMPETE SONICALLY with the 5.1 stuff!!! "Louder is better" just doesn't work when you're in the presence of great masters.

That's right, I predict that the critical mastering engineers of the future will be so embarrassed by the sound quality of the good 5.1 stuff that they won't be able to get away with smashing 5.1 masters. And, hopefully, the two-track reductions that they also remaster (the CD versions) especially if there is a CD layer on the same disc, will be mastered to work at the same LOUDNESS.

In fact, if you tried to turn 5.1 Lyle Lovett, Michael Jackson, Aaron Neville, or Sting into a K-14, they just would sound horrid, on any reasonable 5.1 playback system!

The DK meters, set to K-20 demonstrated clearly that K-20 rules in 5.1. In fact, after a while I simply turned off the peak portion of the meter as it was distracting. So we could watch the VU-style levels and see the techniques used by each of the mix engineers. At K-20 and with 6 speakers running, you have so much headroom that it is hardly necessary to watch the peak meters at all. Furthermore, at 24 bits, there is absolutely no necessity to hit 0 dBFS ANYMORE AT ALL.

The proof is in the pudding, when you try your first 5.1 master you will see clearly what I mean. K-20-style metering and calibrated monitoring becomes a MUST in 5.1.

If you are interested in discussing the ramifications of these topics, please contact the author, Bob Katz.

Credits
Many thanks to: Ralph Kessler of Pinguin for reviewing the manuscript and suggesting valuable corrections and additions.

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Appendix 1: Definition of Terms
Average - "Integrated" level of program, as distinguished from its momentary peak levels.
Average level - Area under the rough waveform curve, ignoring momentary peaks.
Averaging method - (such as arithmetic mean, or root-mean-square) must be specified in order to determine area under curve.
Compression - "dynamic range reduction". Not to be confused with the recent use of the word to describe digital audio coding systems such as AC-3, MPEG, DTS and MLP. To avoid ambiguity, refer to the latter as coding systems, or more exactly, data-rate-reduction systems.
Crest Factor - ratio between peak and average program levels, or ratio of level of instantaneous highest peak to average level of program. There is no standard for the averaging method to be used in determining crest factor. I've used a VU characteristic for purposes of illustration. Unprocessed music exhibits a high crest factor, and a low crest factor can only be obtained using dynamic-range compression.
Headroom - ratio between peak capability of medium and average level of program. There is no standard averaging method for determining headroom. I've used a VU characteristic for purposes of discussion.
Metadata - "data about data" Coding systems such as AC-3, DTS, and MLP can insert control words in the data stream which describe the data, the audio levels, and ways in which the audio can be manipulated. Metadata permits the insertion of an optional dynamic-range compressor located inthe listener's decoder, bringing up soft passages to permit listening at reduced average loudness. The control word dynrng controls the parameters of this compressor in the AC-3 system and hopefully will also be used in MLP. The advantage of this approach is that the source audio remains uncompromised. Other important control words include dialnorm and mixlev.
MLP - (Meridian losslesss packing). The lossless coding system specified for the DVD-Audio disc.
VU meter - According to A New Standard Volume Indicator and Reference Level, Proceedings of the I.R.E., January, 1940, the mechanical VU meterused a copper-oxide full-wave rectifier which, combined with electrical damping, had a defined averaging response according to the formula i=k*e to the p equivalent to the actual performance of the instrument for normal deflections. (In the equation i is the instantaneous current in the instrument coil and e is the instantaneous potential applied to the volume indicator)...a number of the new volume indicators were found to have exponents of about 1.2. Therefore, their characteristics are intermediate between linear (p = 1) and square-law or root-mean-square (p=2) characteristic."

Appendix 2: SMPTE Practice
All quoted monitor SPL calibration figures in this paper are referenced to -20 dB FS. The "theatre standard", Proposed SMPTE Recommended Practice: Relative and Absolute Sound Pressure Levels for Motion-Picture Multichannel Sound Systems, SMPTE Document RP 200, defines the calibration method in detail. In the 1970's the value was quoted as "85 at 0 VU" but as the measurement methods became more sophisticated, this value proved to be in error. It has now become "85 at -18 dB FS" with 0 VU remaining at -20 dBFS (sine wave). The history of this metamorphosis is interesting. A VU meter was originally used to do the calibration, and with the advent of digital audio, the VU meter was calibrated with a sine wave to -20 dB FS. However, it was forgotten that a VU meter does not average by the RMS method, which results in an error between the RMS electrical value of the pink noise and the sine wave level. While 1 dB is the theoretical difference, the author has seen as much as a 2 dB discrepancy between certain VU meters and the true RMS pink noise level.
The other problem is the measurement bandwidth, since a widerange voltmeter will show attenuation of the source pink noise signal on a long distance analog cable due to capacitive losses. The solution is to define a specific measurement bandwidth (20 kHz). By the time all these errors were tracked down, it was discovered that the historical calibration was in error by 2dB. Using pink noise at an RMS level of -20 dBFS RMS must correctly result in an SPL level of only 83 dB. In order to retain the magic "85" number, the SMPTE raised the specified level of the calibrating pink noise to -18dB FS RMS, but the result is the identical monitor gain. One channel is measured at a time, the SPL meter set to C weighting, slow. The K-System is consistent with RP 200 only at K-20. I feel it will be simpler in the long run to calibrate to 83 dB SPL at the K-System meter's 0 dB rather than confuse future users with a non-standard +2 dB calibration point.
It is critical that the thousands of studios with legacy systems that incorporate VU meters should adjust the electrical relationship of the VU meter and digital level via a sine wave test tone, then ignore the VU meter and align the SPL with an RMS-calibrated digital pink noise source.

Improved measurement accuracy if narrow-band pink noise is used
There are many sources of inaccuracy when determining monitor gain when using pink noise. Using wideband (20-20 kHz) pink noise and a simple RMS meter can result in low frequency errors due to standing waves in the room, high frequency errors due to off-axis response of the microphone, and variations in filter characteristics of inexpensive sound level meters. For the most accurate measurement, use narrow-band pink noise limited 500-2kHz, whose RMS level is -20 dBFS. This noise will read the same level on SPL meters with flat response, A weighting, or C weighting, eliminating several variables.

For even more accuracy, a spectrum analyzer can be used to make the critical 1/3 octave bands equal and reading ~68 dB SPL, yet totalling the specified 83 dB SPL.

Appendix 3: Detailed Specifications of the K-System Meters
General: All meters have three switchable scales: K-20 with 20 dB headroom above 0 dB, K-14 with 14 dB, and K-12 with 12 dB. The K/RMS meter version (flat response) is the only required meter--to allow RMS noise measurements, system calibration, and program measurement with an averaging meter that closely resembles a "slow" VU meter. The other K-System versions measure loudness by various known psychoacoustic methods (e.g., LEQ and Zwicker).
Scales and frequency response: A tri-color scale has green below 0 dB, amber to +4 dB, and red above that to the top of scale. The peak section of the meters always has a flat frequency response, while the averaging section varies depending on version which is loaded. For example: Regardless of the sampling rate, meter version K-20/RMS is band-limited as per SMPTE RP 200, with a flat frequency response from 20-20 kHz +/- 0.1 dB, the average section uses an RMS detector, and 0 dB is 20 dB below full scale. To maintain pink noise calibration compatibility with SMPTE proposal RP 200, the meter's bandpass will be 22 kHz maximum regardless of sample rate.

Averaging time and Weighting Filters:
The average section of all K-Meters has an integration time of 600 ms and fall time of 600 ms. The filter section of Meter K-20/ITU, K-14/ITU, and K-12/ITU correspond with ITU BS.1770 recommendations for the filter to be used for loudness measurement. Regardless of the frequency response or methodology of the loudness method, reference 0 dB of all meters is calibrated such that 20-20 kHz pink noise at 0 dB reads 83 dB SPL, C weighted, slow. Psychoacousticians designing loudness algorithms recognize that the two measurements, SPL and loudness are not interchangeable and take the appropriate steps to calibrate the K-system loudness meter 0 dB so that it equates with a standard SPL meter at that one critical point with the standard pink noise signal. The RMS calculation should use at least 1024 samples to avoid an oscillating meter with a low frequency sine wave.
Scale gradations: The scale is linear-decibel from the top of scale to at least -24 dB, with marks at 1 dB increments except the top 2 decibels have additional marks at 1/2 dB intervals. Below -24 dB, the scale is non-linear to accomodate required marks at -30, -40, -50, -60. Optional additional marks through -70 and below . Both the peak and averaging sections are calibrated with sine wave to ride on the same numeric scale. Optional (recommended): A "10X" expanded scale mode, 0.1 dB per step, for calibration with test tone.
Peak section of the meter: The peak section is always a flat response, representing the true (1 sample) peak level, regardless of which averaging meter is used. An additional pointer above the moving peak represents the highest peak in the previous 10 seconds. A peak hold/release button on the meter changes this pointer to an infinite high peak hold until released. The meter has a fast rise time (aka integration time) of one digital sample, and a slow fall time, ~3 seconds to fall 26 dB. An adjustable and resettable OVER counter is highly recommended, counting the number of contiguous samples that reach full scale.

FOOTNOTE

The late Gabe Wiener produced a series of classical recordings noting in the liner notes the SPL of a short (test) passage. He encouraged listeners to adjust their monitor gains to reproduce the "natural" SPL which arrived at the recording microphone. The author used to second-guess Wiener by first adjusting monitor gain by ear, and then measuring the SPL with Wiener's test passage. Each time, the author's monitor was within 1 dB of Wiener's recommendation. Thus demonstrating that for classical music, the natural SPL is desirable for attentive, foreground listeners.

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Here's one of the secrets of the mixing engineers
To avoid squashing, if it doesn't sound loud enough to your ears, turn up the monitor! If you find that you've been forced to apply limiting or compression just to keep the meters from overloading, then you've been going about this backwards. Instead, turn down your individual mix levels several dB, then get rid of any compression you were using to "protect" the 2-mix. Now your mix is at a lowered meter level, so turn up your monitor gain to arrive at the same loudness--only this time it won't sound squashed. Leave the monitor at that position as you continue to mix (mark it so you can get back to it).

In 24-bit recording you can make a perfectly good mix that peaks between -3 and -10 dBFS with no loss of quality, in fact, with improved quality. So if the mix gets too loud by your ears, then turn down the elements that are too hot in the mix instead of turning down the monitor again, with no fear of mixing "too low". In other words, a high monitor gain gives you less temptation to overcompress. High monitor gain does not necessarily mean high monitor output from the speakers--it means that the mix level had to be lower. For example, visit the CD Honor Roll and check out the great-sounding Lyle Lovett selection, which is close to the dynamics of a raw mix. Notice that in order to listen to it, you have to turn up your monitor gain. That's approximately where your monitor control for a dynamic raw mix should be sitting (within 4-6 dB) before mastering. Obviously, a lot of today's hypercompressed masters would require turning down the monitor, but we're trying to show you how not to ruin the record in the mix stage (and hopefully not in the mastering, either!).

Know thy monitors
But even when you do, never be fooled. Take your mixes around and listen to them on several other systems that you know; then go back into the control room and if they do not translate, try to adjust your mixes in the areas where they do not translate. HOWEVER, be aware of the extremes. If it sounds reasonably good in a car, for example, don't be tempted to turn up the highs for the car or it will screech (horribly) EVERYWHERE else. First of all, in the mastering we have much more experience in knowing how far to go and make sure that a recording is not made bass-shy just because it sounds boomy in a naturally-boomy car, for example.

Always mix to the highest possible wordlength
Even if the source tracks are 16-bit! Do not sample rate convert. When you're ready to bounce or prepare files, please see our guidelines page for suggestions on making file names and file types.

Track important instruments in stereo
In the days of 8 track you had to be very careful about allocating tracks. But those days are gone. You have enough tracks to splurge now! So there's no reason to conserve on tracks during the tracking stage. The stereo image and depth of your final product will be determined by your skill in mixdown at using delays, reverberation, effects, and your skills in tracking, how you tracked your instruments. Try to make a plan beforehand of how your soundstage might look, where the instruments might be placed. Realize that it probably will not hurt, and probably will help to record your important instruments in stereo.

For example, even a pair of bongos that are destined to be on the right side of the soundstage will sound better if one bongo mike is panned full right and the other somewhat right of center. This is because the ear decodes the natural space and delays picked up by those microphones, actually enhancing their definition in the mix (if the room acoustics are good).

Another example: Electric guitar. Capture the direct to one track. Capture the output of the loudspeaker with a close mike to another track. Capture the medium distant sound of the speaker bouncing from the walls of the room with another mike. Listen to the combination of these sources panned to different places, and also listen in mono to make sure you have not created phase cancellations. By using stereo miking and natural room acoustics in the tracking, and possibly artificial delays and good stereophonic reverberation in the mixing, your mix will sound richer and deeper. Not everything should be tracked in stereo, but don't skimp on elements that will increase the depth and space of your recording. Of course you will need a foreground, middleground, and background in the mix, but it's a lot harder to create a location and space for an instrument if you had only recorded it in mono.

In the mixing, use artificial reverberators that enhance depth and space and do not sound flat, plastic or "cheesy." Use artificial delays to locate instruments in space, not just simple panners.

Levels
Try to not exceed -3 dBFS peak on a peak meter on the highest peak of the mix. Low levels are perfectly acceptable in a 24 bit system. Once you see that the highest peak is in the range of, say, -10 dBFS to -3 dBFS, then from that point on, if you can hear it, the low level passages are ok. Preserve dynamic range! Assume that if anyone is going to ruin the master, let it be me (the mastering engineer). If the mix sounds good, then soft passages automatically are NOT too soft. Of course, if you think a soft passage sounds too soft in the mixing, then of course try to fix that during the mixing. But these can easily be dealt with and often more efficiently in mastering, as we have the context of the album in mind.

If you have a VU meter, use it. With a sine wave, adjust it so 1 kHz, 0 VU is equal to -20 dBFS on the peak meter. Use the VU, ignore the peak, and you'll start making better mixes.

Original sources, please
If at all possible, deliver a generation that is as close to the original as possible. If it's on CD ROM, then cut a CDR directly from your hard disc files. Speed of cutting? Try to use Taiyo Yuden or other reputable blanks, and cut at 4X to 8X speed. These will PROBABLY produce the best results. Murphy's Law: Allow for Murphy. Do not ASSUME that all the files will transfer successfully over here and that the CD-ROMs you have cut are perfect. Allow for the possibility that on the very last minute of the very last hour of the very last day, we may have to go to a backup CD-ROM, or you may have to cut another, because of some error or other problem in the transfer. Do not paint yourself into a corner. Make backups. Do not destroy or erase any source hard disc at the origination studio until the mastering has been completed.

Vocal Up/Down or (Even Better) Gang of Four

A) Make a lead vocal up (1/2 or 1 dB, you be the best judge) version. Do it NOW before you forget. It's a lot easier to do it NOW than to discover in the mastering that you should have. Occasionally do a vocal down (1/2 or 1 dB) version if you think it may be useful; then again, it only takes 3 minutes to do a vocal alt version when you're in the heat of mixing, but it takes forever to try to fix it in the mastering if you forgot.

or

B) Gang of Four - Stems or Splits

Professional mixing engineers never get caught with their pants down when they produce the Gang of Four. This is even better than Vocal Up/Vocal Down and is not any more work.

Here's the key: You produce up to four synced stems. In a time pinch, you can produce only the first two or three.

Stem #1. TV (that’s instrumental plus chorus or background vocals).
Stem #2. Lead Vocal(s) (plus its reverb of course ---basically muting everything else).
Stem #3. Full Mix (that’s what I will use unless there is a problem, and it's also a reference to prove that #1 and #2 were made correctly).
Stem #4. Instrumental (by adding this to #1 we can reduce the chorus level. By subtracting this from #1 we can increase the chorus level. By subtracting this from #3 we can increase lead and chorus. And so on!

Synchronized stems are produced by running a separate mix pass from the same starting sample each time without changing any gains. You mute the tracks that you don't want to hear. That way any reverbs or other processing which were applied on the full mix remain on the stems. Don't be afraid if the vocal-only version has 1 minute of blank at the head, that's part of the design!

In mastering, the sum of Stem #1 and Stem #2 at unity gain = Full mix. If we want lead vocal down, we just take the level of Stem #2 down a hair. And through other combinations we can control instruments or all vocals. In mastering, if a vocal is sibilant, we can apply a de-esser just to the vocal track, which is less of a compromise than de-essing the full mix. If a bass instrument needs to come up, we can equalize the instruments without making the vocal any bassier. And so on.

Many mix engineers argue (correctly) that if they are using bus compression the stems will not reflect the same sound they got on the full mix. This is true, if you are doing strong bus compression, the interaction between the peaks of the mix and the individual elements will not be the same when using stems. In that case, a legitimate gang of four cannot be produced!

The gang of four protects you and your clients in many ways. It gives you archive options and alternative options. It gives you the TV mix the client forgot to ask for but requests six months later! It allows cleaning up dirty words without dropping the music out. Develop the discipline to do the gang of four. You won't be sorry!

When, Why, and How to Make Stems
I've definitely reached the conclusion that the less compromise you can make in the mastering process, the better the result. mixing is mixing and mastering is mastering, and you should make the very best, finished mix that you can before sending it to mastering. Stems in mastering are not meant to be a substitute for a bad mix, nor a "mix fixer", but they can be used in the mastering process to polish or aid a very good mix with less compromise than a full mix. The case of vocal up/down is a clear one where the mix is otherwise preserved, but the mastering engineer can tweak the lead vocal if the mastering processing seems to affect the vocal/instrument relationship, or the producer thinks twice about the vocal level, or some word somewhere needed to be punched but someone forgot. The mastering engineer is not a closet or frustrated mix engineer, we treat the stems as safeties or convenient means to an end, not as infinite "remix opportunitites".

In more severe cases, you may have produced an otherwise great mix, but you had a little trouble with your monitoring and produced too little bass instrument, too much kick, and the lower midrange is a little bit muddy. This is a potentially bad (not lethal) combination for mastering and if the client has time, I recommend a remix. But in situations when time is tight, I have asked the client for stems, and the results have ALWAYS been better than if I had mastered from the combined two-track.

In mastering, the question comes whether I should first remix the stems in a remix session, then master-----or remix and master in one step. The more stems, the more it becomes a true remix, and the more inclined I would be to put on my mixing hat and mix first, then master. But with only 3 to 6 stereo stems, I find that I can get excellent results mastering with "supplementary mixing" going on. I'm mostly concentrating on mastering, using the stems to aid in the goals of mastering without compromise, such as using the vocal stem when de-essing is needed instead of de-essing and compromising the entire mix. Or equalizing the bass instrument instead of using overall bass frequency eq on a full mix, which has less compromise on the vocal, keyboards, etc. Of course, this kind of precision is usually not needed, and I master from the full mix more than 90% of the time. For example, the mastering processing is going to affect the clarity of the midrange and through "slop" will probably leak down into the bass region, hopefully for the better. But in the case of this lopsided mix I just cited, the mastering processing could easily make one range better while making the other worse.

Is this idea of mastering from stems heresy? It's certainly a dangerous technique if placed in the wrong hands. You can end up with a less than ideal mix or less than ideal master if the mastering engineer does not think holistically. But if placed in the hands of an experienced mastering engineer, I think mastering from stems can produce a subtly better product or do it more efficiently. Producing stems the first time in the mix session reduces the number of calls for a costly and inconvenient remix! An ounce of prevention is worth a pound of cure.

Click here to see what other engineers had to say on the subject of Compression in Mixing.

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Copyright Digital Domain, Inc. We invite you to link to our site, which will be periodically revised.

{slide=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".{/slide}
{slide=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.
  {/slide}

{slide=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.

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.{/slide}

{slide=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.{/slide}

{slide=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.{/slide}

{slide=VI. No Longer The Missing Link - Affordable 24-bit file interchange}
At Digital Domain, 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.{/slide}

{slide=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!{/slide}

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Please visit our Delivery & FTP page.

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For the Secrets of the mastering Engineer article, which I wrote for TC Electronic to enclose with their Finalizer and other processors, click here and scroll to the article "Secret of the mastering Engineer or browse some of the other excellent articles and information at the TC website.

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  Copyright Digital Domain, Inc. We invite you to link to our site, which will be periodically revised.

Accurately Set Up a Subwoofer With (Almost) No Test Instruments

Bass frequencies are extremely important to sound reproduction. Everyone is interested in getting their bass right, but most people haven't a clue how to proceed. This article will help to settle the process of integrating an active subwoofer with an existing "satellite" system. If your room and loudspeakers are good, you'll only need two test CDs and your ears to adjust your subwoofer. If your room is not so good, or you want to refine the sound even further, then we'll discuss the best way to integrate test equipment measurements with your hearing. The simple listening test will also reveal if your room has problems and if it's time to hire an acoustician.

Let's review the basic requirements for smooth, extended bass response.

Conquering the Room
Many people are proud of the "ideal dimensions" of their listening room. In general, the larger the room, the fewer audible problems with low frequency standing waves (nodes and antinodes). To get smooth and even bass requires ceilings taller than 10 feet, width greater than 12 feet, and length greater than 25 feet (30 or more for deep bass). Dimensions (including diagonals) should not be multiples of identical wavelengths, to avoid buildup at octave resonances. Of course, larger rooms may need absorption to keep the reverberation time down, but standing waves don't tend to build up awkwardly in larger rooms. It's also important to use absorption so that the decay time at low frequencies is roughly similar to that at mid and high frequencies. This is called a "neutral room."

Lightweight, flexible walls act as diaphramatic absorbers, where some bass frequencies will escape out the walls, never to return. In my opinion, the ideal is a solid concrete (block) wall, but proper construction with plaster lathe, wood, and/or double sheet rock can accomplish similar results. But solid walls create problems of their own; a world-class room usually requires some absorption and/or diffusion to deal with resonances and echos. Watch out for cavities within the walls, which can cause resonances. Creating a large room with good bass response, interior acoustics, and outside isolation, is the role of a professional acoustician. This article will share some secrets in the fine tweaking of systems in good rooms; don't dream of building a room from scratch without hiring an acoustician.

Speaker Mounting - Spikes or Isolators?
Soffit-mounting involves recessing loudspeakers into a cavity in the wall, with the edge of the loudspeaker flush to the wall. Soffit-mounting requires the expertise of an experienced acoustician, and is beyond the scope of this article. The main loudspeakers must be decoupled from the floor. Heavy, rigid stands should have a top no larger than the bottom of the loudspeaker to avoid diffraction (a form of comb filtering). I've had great success spiking speaker stands (using spikes, or "tiptoes") through holes in the carpet. Some authorities recommend a damping pad underneath a heavy, full range speaker instead of spikes. Whichever mounting method, the goal is to reduce sympathetic vibrations or traveling waves in cabinets, floor and walls. The resonant frequency of the box and stand should be extremely low. Hit the box with your fist and confirm it does not have a resonant character; sweep a sine wave through the system and listen for vibrations. I've had great success with a very thin isolator (Dr. Scholl's) between the speaker and the stand which compresses almost completely under the speaker's weight.

Listener position
If you're sitting in an antinode, there's always going to be a dip at that frequency, and no amount of equalization will correct the acoustic problem.

Speaker position
Ironically, solid walls aggravate the interaction of loudspeaker position and frequency response. The closer the loudspeaker to walls and especially corners, the greater the bass level. You may have the "smoothest," most accurate satellite (main) speakers in the world, but they must be positioned to avoid side wall reflections and must be far enough from all walls to reduce resonances.

Near Field Monitoring?
I wouldn't master with near-field monitors, but I will mix with them. Near-field monitoring was devised to reduce the effects of adverse room acoustics, but if your room acoustics are good, then "Mid-field" or "Far-field" will provide a more accurate depth and spatial picture. There must be an obstruction-free path between the monitors and the listener. What is the biggest impediment to good sound reproduction in a recording studio? The console. No matter how you position the monitors, the console's surface reflects sound back to your ears, which causes comb filtering, the same tunnel effect you get if you put your hand in front of your face and talk into it. Or if you wear a wide-brimmed hat, which produces an irregular dip around 2 kHz. It amazes me that some engineers aren't aware of the deterioration caused by a simple hat brim! Similarly, I shudder when I see a professional loudspeaker sitting on a shelf inches back from the edge, which compromises the reproduction. The acoustic compromise of the console can only be minimized, not eliminated, by positioning the loudspeakers and console to increase the ratio of the direct to reflected path. Lou Burroughs' 3 to 1 rule can be applied to acoustic reflections as well as microphones, meaning that the reflected path to the ear should ideally be at least 3 times the distance of the direct path.

What about measurements?
Can't we just measure, adjust the crossovers and speaker position for flattest response, then sit down and enjoy? Well, since no room or loudspeaker is perfect, measurements are open to interpretation, and frequency response measurements will always be full of peaks and dips, some of which are more important to the ear than others. Which of those many peaks and dips in the display are important and which ones should we ignore?
I've found the ear to be the best judge of what's important, especially in the bass region. The ear will detect there's a bass problem faster than any measurement instrument. The measurement instrument will help to pinpoint the specific problem frequencies, whether they're peaks or dips, and by supplying numbers, aid in making changes. The whole process is very frustrating, and it's inspired my search for setup and test methods that use the ear. A perfect setup still requires a multistep process: listen, measure, adjust, listen again, and repeat until satisfied, but it's possible to streamline that process. Here's a listening test for adjusting subwoofer crossovers that uses simple, readily obtainable and cheap test materials, and that's generally as precise as most more formal measurement techniques! If you're setting up a permanent system, dedicate a day to the process; even the easy doesn't come easy. Some brands of subwoofer amplifiers have all the controls or connectors you need; you may have to adapt the process described below to your particular woofer system.

Polarity is not Phase
This is still a confusing topic, perhaps because people are too timid to say polarity when they mean it. The polarity of a loudspeaker refers to whether the driver moves outward or inward with positive-going signal, and can be corrected by a simple wire reversal. Remember that phase means relative time; phase shift is actually a time delay. The so-called phase switches on consoles are actually polarity switches, they have no effect on the time of the signal! Sometimes this is referred to as absolute phase, but I recommend avoiding the use of the term phase when you really mean polarity. If two loudspeakers are working together, their polarity must be the same. If they are separated by space, or if a crossover is involved, there may be a phase difference between them, measured in time or degrees (at a specific frequency). I have a pair of Genesis subwoofers with separate servo amplifiers. There are three controls on the crossover/amplifier: volume (gain), phase (from 0 to 180 degrees), and low pass crossover frequency (from 35 Hz to over 200 Hz). Notice there is no high pass adjustment. The natural approach to subwoofer nirvana assumes that your (small) satellite loudspeakers have clean, smooth response down to some bass frequency, and gradually roll off below that. It's logical to use the natural bass rolloff of the satellites as the high pass portion of the system and to avoid adding additional electronics that will affect the delicate midrange frequencies. So we use a combination of lowpass crossover adjustment and subwoofer positioning to fine-tune the system.

A good subwoofer crossover/amplifier usually provides more than one method of interconnection with the satellite system. The best is the one which has the least effect on the sound of the critical main system. I prefer not to interfere with the line level connections to the (main) power amp feeding the satellites. If your preamplifier does not have a spare pair of buffered outputs, I recommend using the speaker-level outputs of the main power amp. The Genesis provides high-impedance transformer-coupled balanced inputs on banana connectors designed to accept speaker-level signals. Connect the main power amp's output to the sub amp's input with simple zip cord with bananas on each end. No real current is being drawn, so wire gauge does not have to be heavy. Double-bananas make it easy to reverse the polarity of the subwoofer, a critical part of the test procedure. Some subwoofers use a 12 dB/octave crossover, others 18 or more. Interestingly, for reasons we will not discuss here, a 12 dB crossover slope requires woofers that are wired out of polarity with the main system. My new sub crossover uses a 24 dB lowpass slope, which also requires polarity reversal, but to make it easy on the mindless, the internal connections are reversed, and you're supposed to connect "hot to hot" between the main power amplifier and woofer amplifier. Leave nothing to doubt-we must confirm the correct polarity. Steep slopes like 18 and 24 dB are good choices to get the subwoofer to roll off before it interferes with the midrange response.

You have to sit in the "sweet spot" for the listening evaluation. If your subwoofers have an integrated amplifier, you'll need a cooperative friend to make adjustments. Since the Genesis amplifier is physically separate, I was able to move the subwoofer amplifiers to the floor in front of the sweet spot, and make my own adjustments. Here are the two test CDs:

1. The Mix Reference Disc, Deluxe Edition, MRD2A. Since this disc is now out of print, Mix Editor George Petersen has kindly given me permission to put the test tones up on our site so you can make your own custom Subwoofer test CD. You can also print a traycard from the PDF file I've provided. You can also use any source of 1/3 octave filtered pink noise.
2. Rebecca Pidgeon, The Raven, Chesky JD115, available at record stores, high-end stereo stores, or from Chesky Records.

I recorded Rebecca's disc in 1994. Track 12 is Spanish Harlem, which has a slow, deliberate acoustic bass part that makes it easy to identify notes that "stick out" too far and covers the major portion of the bass spectrum. This record has never failed to reveal the anomalies of different rooms and loudspeakers in several years of use as a musical reference. The ear is better with instant comparisons than absolute judgments, and this test relies on our ear's ability to make comparisons. All musical instruments and transducers produce harmonics as well as fundamentals. To the best of our ability to discriminate, we will be concentrating on the fundamental tones in this piece of music. If your loudspeakers have significant harmonic distortion, they can complicate or confuse the test. Many studio loudspeakers are designed for high power handling at the expense of tonal accuracy or distortion. This test is not for them. If you want accurate bass, it's time to replace the loudspeakers and probably hire an acoustician with a distortion analyzer.

Start by evaluating the satellite system with the subs turned off. Listen to the bass at a moderate level equal to or slightly louder than the natural level of an acoustic bass. Listen for harmonic distortion: if it doesn't sound like a "transparent" acoustic bass, fix the problem with the satellites, first. Listen for uneven notes. If the lower note(s) of the scale are successively softer in level than the higher notes, then you have a perfect candidate for a subwoofer. If intermediate bass notes are weak or strong (uneven bass), the satellite loudspeakers may be too close to the corners, in a node or antinode, the listening position may be in a standing wave, or the satellites themselves poorly designed. It may be time to bring in an acoustician. But if the satellite bass is even, you can move on to the next step, adjusting the subwoofers.
Spanish Harlem, in the key of G, uses the classic 1, 4, 5 progression. Here are the frequencies of the fundamental notes of the bass. If your loudspeaker has sufficiently low harmonic distortion, it will not affect your judgment of the power of the bass notes, which are already affected by the natural harmonics of the instrument.

		49  62      73          65      82      98            73      93      110
	

As you can see, this covers most of the critical bass range. If the lowest note(s) is weaker than the rest, then you are a candidate for a subwoofer. My satellites behave in the classic manner, with the lowest note (G, 49Hz) slightly low in level, but the rest fall in a balanced line. I've been in small rooms where one or more of the intermediate notes are emphasized or weak, which suggests standing wave problems. Repositioning the satellites may help. Avoid equalization, which is a nasty band aid...proper acoustic room treatment is the cure. You could conceivably add a subwoofer out of phase at the frequencies in question, but that's a technique that should remain confidential between you and your analyst. Fix the acoustic problems first and you'll be happier.

If your satellite system passed the initial examination, next step is to decide on a starting (approximate) subwoofer location. A satellite-subwoofer system has tremendous flexibility, offering in theory the best of two worlds. The satellites can be placed on rigid stands at ear level, far from corners and side walls, reducing floor and wall reflections and comb-filtering in the midband. And the subs can be placed on the floor, in the position that gives the most satisfactory bass response, integrated with the satellites. If you only have one (mono) subwoofer, start by placing it in the middle between the stereo speakers. Contrary to popular belief, stereo subwoofers are important, they can improve the sense of "envelopment", the concert hall realism that bass waves are passing by you. Authorities are split on the issue whether a mono or stereo subwoofer setup is more forgiving of room modes. I prefer the sound of stereo subwoofers. A complete discussion of how to place the satellites would require another article, but let's start by saying that you may have to deal with reflections from the side walls by placing absorbers in critical locations. Consider consulting a competent acoustician.

Assuming your satellite system passes the listening test, it's time to find the right crossover frequency, phase and woofer amplitude that will just supplement the lower notes of the scale. Start by placing the subwoofers next to and slightly in front of the satellites. First we must determine the proper polarity for the subwoofers. If your system uses XLR input connectors, build a polarity reversing adapter for this part of the test. This is easier with only one channel playing. Put on the MRCD with full bandwidth pink noise, at a moderate level (70-80 dB SPL). Adjust the crossover to its highest frequency, the phase to 0, and turn up the subwoofer gain until you're sure you can hear the woofer's contribution. Reverse the polarity of the sub. The polarity which produces the loudest bass is the correct polarity. Mark it on the plugs, and don't forget it!

Next comes an iterative process ("lather, rinse, repeat until clean"). Here's a summary of the four-steps: (1, 2, & 3) Using filtered pink noise, we'll determine the precise phase, amplitude and crossover dial position for any one crossover frequency. (4) Then we'll put Rebecca back on and see if all the bass notes now sound equally loud. If not equally loud, then we'll go back to the filtered pink noise and try a different crossover frequency. We keep repeating this test sequence until the bottom note(s) has been made "even" without affecting the others. With practice you can do this in less than half an hour. Adjust each subwoofer individually, playing one channel at a time.
And now in detail:

1) Crossover frequency (lowpass)
Play filtered pink noise (or the Mix CD's multifrequencies) at your best guess of crossover frequency, say 63 or 80 Hz. Notice that the signal has a pitch center, or dominant pitch quality. If the subwoofer is misadjusted, adding the sub to the satellites will slide the pitch center of the satellite's signal. Reverse the sub's polarity (set it to incorrect polarity). With the sub gain at a medium level, start at the lowest frequency, and raise the frequency until you hear the dominant pitch begin to rise (literally, the center "note" of the pink noise appears to go sharp, to use musical terms). Back it off slightly (to a point just below where the pitch is affected), and you have correctly set the crossover to this frequency. Recheck your setting. That's it.

2) Phase
The sub should always be on a line with or slightly in front of the satellite. With the woofer a moderate amount in front of the satellites, the phase will generally need to be set something greater than 0 degrees. Return the sub(s) to the correct polarity. Play the same frequency of filtered noise and increase the amount of "phase" until you hear the dominant pitch rise. Back it off slightly, recheck your setting, and that's it.

3) Amplitude
The subwoofer's settings are exactly correct when its amplitude is identical to the satellite's at the crossover frequency. The subwoofer gain is the easiest to get right because there will be a clear center point, just like focusing a camera. Play the filtered noise, and discover that the pitch is only correct at a certain gain, above which the pitch goes up (sharp), and slightly below which it goes down (flat). "Focus" the gain for the center pitch, which will match the pitch of the satellites without the sub. Recheck your work by disconnecting and reconnecting the sub. The pitch should not change when you reconnect the sub, otherwise the gain is wrong. To be extremely precise, increase the gain in tiny increments until you find the point where the pitch rises when the sub is connected, then back the gain off by the last increment. This process is extremely sensitive.

4) Rebecca
Play Spanish Harlem again. If all the levels of the bass notes are even, you're finished with steps 1-4. If you hear a rise in level below some low note, then the crossover frequency is too high and vice versa. Do not attempt to fix the problem with the subwoofer gain, because that has been calibrated by this procedure, which leaves nothing in doubt except the choice of crossover frequency. Go back to step one and try again. Once all the notes are even, your crossover is perfectly adjusted. Write that frequency down. Then, for complete confidence, check the nearest frequency above and below (go back through steps 1-4), proving you made the right choice. This piece of test music is sufficiently useful that there will be a clear difference between each 1/3 octave frequency choice and it will be comparatively easy to determine the winner. The trick is not to rely on our faulty acoustic memory, but on the ear's ability to make relative comparisons.

More Refinement
Fine tuning the stereo separation (space between the woofers)
If you have stereo subwoofers, their left-right separation must be adjusted. Play Spanish Harlem. Listen to the sound of the bass with the subs off. It should be perfectly centered as a phantom image and and its apparent distance from the listener should subtend a line between the satellites. If it is not perfectly centered or its image is vague, the satellites are too far apart. Now add the subwoofers. The bass should not move forward or backward, and its image should not get wider or vaguer. Adjust the physical separation of the subwoofers until the bass image width is not disturbed when they are turned on. This "integrates" the system. Go back to step one, recheck the amplitude and phase settings for the new woofer position. Everything is now spot on.

Congratulations, you've just aligned a world-class reproduction system! A subwoofer should not call attention to itself, either by location or amplitude. When you play music, the combination of the sub and mains will sound like a single, seamless source.

Now, after logging your settings, sit back, listen and enjoy. You've earned the time off. Don't let anyone touch those hard-earned adjustments, for you can be confident that they are about as good as they're going to get. Play several of your favorite recordings, and listen to the bass. The bass on the best recordings will be acceptable on your reference system; the worst recordings will have too much or too little bass. Now you can be reasonably sure the problem is in the recording, not your room or woofers. What a nice feeling!

How The Pitch Detection Method Works
The 1/3 octave pink noise signal (or the multitone test signal) contains a narrow band of frequencies, whose dominant level is at the center of the band. Thus, you perceive a "pitch" to the signal. When you add a second loudspeaker driver (the subwoofer) driven by the same signal, if the woofer's output does not exactly match the level and distribution of frequencies produced by the main loudspeaker, there will be a shift in the dominance of the multifrequencies, either towards the high end of the band or the low end, perceived as a pitch shift. When the two signals are well-matched in level, freqeuency distribution and phase, you will hear a delicate increase in level, but no change in pitch. By simple comparative listening, taking the woofer in and out of the circuit, you have confirmed that your drivers are matched at the crossover frequency, and that the wavefronts of your main speakers and subs are aligned at the critical crossover frequency.

Of course, we're making certain assumptions...that:

• your satellite system is well designed, linear and rolls off below some defined frequency.
• your subwoofer system is linear and rolls off above some defined frequency.
• the slopes of the two rolloffs are compatible and will integrate.

Your degree of success depends on how closely the two systems meet those requirements.

What To Do When the Results are Less Than Perfect
When interpreting Spanish Harlem, don't get too hung up on little "dips" in level. Dips are less objectionable to the ear than peaks. First, attack problems with resonant notes and then look at the dips. Everything may not be rosy the first time around. Supposing that the subwoofer helped the bottom note(s), which means the crossover is at the right frequency, but some upper note in the progression has been affected. This means the subwoofer position is not optimized, or the subwoofer has some frequency response anomaly. As the sub is moved towards the room corners, the low bass response goes up, previous dips become peaks. There's cancellation/reinforcement between the subs and the satellites, which changes complexly as the sub is moved. Thus, adjusting the subwoofer position is a powerful method to even out the bass, but this type of trial and error is too complicated without test equipment. You could slide the woofer slightly, adjust the crossover as above, listen, move it again, readjust, and listen, but our acoustic memory is too short to tell when we've hit the perfect spot.

Advanced Techniques
Integrating the Instruments with the Ears
Here's where it gets complicated. If you are having problems with uneven bass, we can no longer rely strictly on our ears. If you're comfortable with measurement instruments, then let's proceed. First, listen to Rebecca and mark down the problem frequency or frequencies, either peaks or dips. You'll use that knowledge when you bring in the big guns, the 1/3 octave analyser. The good thing is that Rebecca has already told you where the problems are, so you'll know how to separate the forest from the trees in the 1/3 octave display. I used Spectrafoo (an excellent analysis program for the Mac) in transfer function mode with wide band pink noise into both satellites and subs (one channel at a time). Spectrafoo time aligns the stimulus and response, which helps to separate direct from reflected sound, more accurately representing what the ear hears. Spectrafoo revealed a rising response in my room below 40 Hz, and more important, a little dip in the combined response circa 63 Hz which corresponded with my perception that note was perhaps a little weak. By moving the sub around very slightly and watching the display, I was able to exchange the weakness against the surplus without aggravating any other peaks.

The strength of this method is we're continuously integrating our powerful (almost objective) listening judgments with the "over-powerful" analysis tool. We're using the analyser for general trends, not absolute amplitudes; that's what I mean by separating the forest from the trees. The position of the test microphone should be in the exact listening position. Wear earplugs to keep your ears fresh when you're not required to listen. After moving the woofer, don't forget to readjust the crossover gain and phase with our listening technique.

If all goes well, Spanish Harlem will be even better adjusted and we can rest assured that our system is really really tweaked. Now sit back and enjoy. Oops, your work is never done. Now that you've adjusted your system, I'll let you in on one more secret: Servo amplifiers have internal adjustments that affect woofer damping, make the bass "tighter" or "looser." but that's another story.

Acknowledgments:

Jon Marovskis of Janis Subwoofers introduced me to the concept of a pitch detection technique many years ago. This article refines and expands on his original idea.

Many thanks to Dave Moulton for insightful technical and editorial comments.

Also, thanks for manuscript review and suggestions by Johnson Knowles of the Russ Berger Design Group, Eric Bamberg, Greg Simmons and Steven W.Desper.

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Acoustician Johnson Knowles suggests a viscoelastic polymer pad material like EAR Isodamp C1002 or C1000. The internal damping characteristics ofthe viscoelastics are exceptionally effective as a speaker to stand interface material.

U.S. Consultant Steve Desper recommends STIK-TAK by Devcon Corporation, available at your local hardware store. It's a cheap solution and works well. Australian Greg Simmons has found a similar product--marketed as Blue Tak: "Use enough of it relative to the weight of your speakers. For a small monitor weighing just over 20kg, I used four balls about 15mm in diameter (one under each corner). With 20kg on top of them, these balls squashed down to about 4mm or 5mm thickness, and held the monitor very firmly."

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THE SOUND OF LIFTOFF!

Thanks to the help of a friend at NASA, I was able to record the liftoff of the Space Shuttle Discovery on March 8, 2001 from the Kennedy Space Center VIP viewing site at 3.1 miles from the launchpad (as close as they let anyone get during the launch except for crew members). Also with the courtesy and help of Gary Baldassari, Mike Morgan, and Andy DeGanahl, who supplied some of the equipment used. Andy and I braved the all-nighter and captured the launch at 6:42:09.059 am EST.

{mov}Launch_At_Dawn_320x240_17mb{/mov}
10/20/06. We are very pleased to present to you a quick time video (click here to download) of the shuttle launch with 16 bit 48 K STEREO Apple Lossless Sound. PC Users: If the movie will not run when you click on it, then download Quicktime 7. The music is provided by Orlando's superb Sovereign Brass, whose album I mastered. Audio engineer Andy DeGanahl took the images with his trusty 8mm handicam, Andy, don't quit your day job!

9/10/09. We are very pleased to present to you the Shuttle Launch in SURROUND, at 96 kHz/24 bit. This is a free download of the original 4.0 recording (just the shuttle launch, without the music accompaniment). Thanks to Chris of airwindows.com for inventing the asynchronous conversion that allowed me to synchronize the front and surround channels of this recording, which were made on two independent Masterlink recorders. The result is incredibly realistic and natural. If you have a properly bass-managed surround system with high headroom subs (preferably a pair of subwoofers with accurate response) then this recording will knock your socks off. The proper reproduce gain is +7 dB above the Dolby standard calibration. That's right, turn up your monitor gain 7 db higher than the SMPTE standard 83 (or "85") level calibration. The free audio files are only available to Digido.com registered users (it's easy to register, just press the link at the top left of this page). After registering, click on the MEDIA link at top and choose DOWNLOADS. The discrete PCM WAV files can be found in the GENERAL section of DOWNLOADS. OR---you can burn a DVD-A from the Iso image file, also found in the GENERAL section of DOWNLOADS. If you download and can play these files please go the Contact Us page and give me some feedback! Enjoy--- Bob Katz.

Technical specs of the recording
Four microphones and two independent hard disc recorders at 24 bits, 96 kHz were used, which were sync'd up later to produce a fantastic surround recording. Two spaced omnis at 6 foot left and right distance were DPA 4041s, and on the same stands, "synchronous" Sennheiser MKH-30 figure 8's. When decoded via dual MS decoders to surround, the outdoor enthusiastic audience should subtend an angle from about 45 degrees left or right all the way around and behind the listener, with the NASA announcements to the right and behind you. The shuttle liftoff commands stage front center, but with doppler waves and echos throughout the front soundstage and distant echoes behind you and the audience largely behnd you. Playing this back in surround is a true "environmental experience."

On the Spectrafoo Audio Analysis
Through the magic of Spectrafoo's audio analysis tools, the audio "portrait" below demonstrates that there's nothing like being there. The spectragram runs from T minus 4 seconds to about T plus 2 minutes. I don't think there's anything on earth that compares with the sound and sight of that fire-breathing monster on liftoff. If you study these incredible specs, including a spectragramic timeline of the liftoff, you will see that to do justice to the experience, you will need a low-distortion subwoofer system capable of producing up to ~119 dB SPL on peaks at 25 Hz and ~116 dB SPL at 16 Hz and below! If not, then you will not be able to feel the chest-thumping, clean solid bottom that is produced. Ironically, the shuttle liftoff from the VIP site is "just loud enough" in person, a pleasant and not ear-damaging experience. Think of it as an 8.3 GWatt amplifier/loudspeaker with zero percent distortion and response down to DC! Running at say, 40% efficiency, that would take 20 thousand megawatts from the breaker box! Those figures are calculated by Dick Pierce from the comparable Saturn 5 moon rocket. These are the figures at 0 foot distance. Of course, some power has been dissipated at 3.1 miles, but examine the astonishing figures below.

It turns out an LFE channel is not needed to reproduce this recording with a properly bass-managed system, because "bass" is the focus of this recording and it requires turning up the gain by 7 dB ABOVE Dolby standard. However, it could not be engineered to reproduce at Dolby Standard monitor gain without redirecting the bass to an LFE channel or it would overload the digital level. But then the bass would not be in stereo, and the shuttle definitely sounds better to me with stereo bass.

Thanks for reading!

Bob Katz

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Copyright Digital Domain, Inc. We invite you to link to our site, which will be periodically revised.

This is an excellent, entertaining, short video by Matt Mayfield that explains the problem with the loudness war in a couple of minutes. Suitable for lay people, musicians, engineers! (This is a high-resolution version so please have patience for the transfer).

{mov}Loudness_War-small{/mov}
Click here to download it! (on a fast cable modem, this one loads and plays immediately. For slower internet connections, you may have to wait for the Quicktime video to appear). Try to listen on a pair of fairly decent speakers. I have a pair of Realistic Minimus 7 speakers hooked to an NAD 30 Watt amp and my Ibook, and the point gets across very well!

Thank you, Matt, great production job!

Threads from the Mastering Webboard

From: K.K. Proffitt Date: Monday, June 09,2003 05:08 PM

What I say is mix at a comfortable level and check at a very
quiet level and then at the standard level (and we should all be on the same
page with 83, 84 or 85 dBSPL for large room--but let's not go there and assume
85 for now, op. cit. RP200-2002). I have to tell you that about 50% of the
movies released are too loud at a cal'd level of 85 dBSPL for me to listen to,
but I check everything I send out at that level and have a feel for what is
expected. Bulla's research shows that engineers are mixing too damn loud. They
actually show serious threshold shift starting with Monday and getting worse
until Thursday! Repetitive threshold shift leads to high-end hearing damage, so
I think we should not suggest that mixers mix as loud as they want, but that
they keep an SPL meter at the desk and learn to LISTEN at reasonable levels.
Regards, KK Proffitt chief audio engineer, JamSync, Nashville
http://www.jamsync.com

Well, I'd certainly pick "Joshua Judges Ruth" by
Massenburg (artist: Lyle Lovett) over the Big Band one, but good idea anyway.
As an example CD to use to set your monitor gains. Lyle Lovett CD...

-----------------------------------------------

Conf: NEWS, YOU CAN USE From: Dennis Whitney
Date: Tuesday, June 10, 2003 10:08 AM

There is the root of the
whole problem. Is compressing the 2 bus a band aid for impatient mix engineers?
When I was first learning, the engineer that educated me said that compression
was to be used sparingly so that the dynamics were unaffected, use the faders
to mix and not the rack. I've spent over 13 years with that in my head. I don't
have a compressor patched or inserted on the 2 buss. I can understand when its
used for adding flavor to fit a genre much like reverb, but it seems there are
a lot of mixers who because of the lack of mentorship or education use the
tools for ways they weren't intended. I say intended because the older
recording I have and I'm sure your aware of it to, do not exhibit compression
levels near to todays records. If all the mixers are referencing squashed cd
nobody is going to learn anything. Another problem I see is deaf mixers. The
monitoring in most places hurt my ears. How many of you played in band without
ear protection, and how does that effect your tolerance or sensitivity to
levels. My dad in his earlier years as an E/E did a few years worth of work at
Boeing Sonic Test Facility, he damaged his ears by not following procedures.
When I was growing up, he always taught me to protect my ears and not listen to
things too loud. Every band I was ever in I used plug for my ears. How many of
you have had their ears tested and frequency sensitivity plotted? Overlay that
curve with the mixer's curve, throw in the curve differentials of the rooms and
you see lots of problems in perception. Mixer using cd like Audioslave as
references will lead to more crushed cd, more deaf ;) mixers, worse mixes..on
and on and on until something snaps. Maybe it will get so bad radios can't play
the music anymore. Either that or people start taking advice from professional
audio engineers who understand not only the history of the processes, but the
future of the music. Long term vision verses short term profit. Thanks for
listening. Peace, Dennis

-----------------------------------------------

From:Bob Katz
Date: Tuesday, June 10, 2003 12:30 PM On 6/10/03 10:08:00 AM,
Dennis Whitney wrote: There is the root of the whole problem. Is compressing the 2
bus a band aid for impatient mix engineers? When I was first learning, the
engineer that educated me said that compression was to be used sparingly so
that the dynamics were unaffected, use the faders to mix and not the rack

That's the way I always worked. I don't think anyone had to teach me that. I
just learned from my new Full Sail intern that they're TEACHING their students
to put a compressor on the 2-bus automatically. Very tragic. BK

-----------------------------------------------
From: Ronny Morris Date:Tuesday, June 10, 2003 01:09 PM

Hypercompression in mastering pop and modern rock is the norm nowadays, doesn't
make sense in teaching them the norm's when
there are better ways to do it. You can't diss the ME's for using too much
compression and not recognize that there is a big problem and growing from the
mix engineers that are doing the exact same thing. Teaching a mix engineering
student to automatically compress the 2 buss, is no different than ME's that
you have pointed, out that automatically hypercompress the material without the
client requesting it. Two norms that most music would be better off by
re-establishing acceptable procedures. I feel that abuse of compression in
mixing_or_mastering ruins music more than it helps it. What difference does it
make if it's the mix engineer or the mastering engineer that kills the
dynamics? The song is 6 feet under regardless. Ronny Morris

-----------------------------------------------

From: Ronny Morris Date:
Tuesday, June 10, 2003 04:46 PM

The interactions from a kick drum on the buss
compressor cannot be mimicked by channel compression. Please explain exactly
what you can do to a bass drum when the compression that you add to it, is in
the domain of the other instruments and effecting them as well, that you can't
completely do when the kick is isolated to it's own track? To the contrary, you
have more control of compression on a kick drum, when the compressor is
reacting_only_to it and not to higher level frequencies and peak transients
introduced from the rest of the instruments in the two buss mix, where longer
release times are beneficial for low end and shorter release times are
beneficial for quick transients and continous levels.  Besides, you're
automatically assuming everyone (or most) suck. I assume the opposite. I'm not
assuming that most people suck, this isn't a problem with every mix that I
receive, however it continues to grow on the mixing level. It's taken long
enough for folks that frequent this board to realize the problem of
compression/limiting abuse and to voice together and try to bring back the life
to the music, through education, but it's just hitting the mix engineers on the
project studio level and will get worse. Plain and simple Brad, they are
learning how to squash the material too. If the client doesn't want dynamic
range, you still have more control adding it in mastering, rather in mixing and
when it's in the digital domain, the less processing the better, most of the
time.

-----------------------------------------------

From: Brad Blackwood
Date: Tuesday, June 10, 2003 05:10 PM

Please explain exactly what you can do
to a bass drum when the compression that you add to it, is in the domain of
the other instruments and effecting them as well, that you can't completely
do when the kick is isolated to it's own track? Ummm, may be the fact that
the bass and drums will drive the compressor, dragging on the gtrs, overheads,
vox, everything? That CANNOT be accomplished with I/O compression, regardless
of what you think. Believe it or not, there are actually really good, smart
engineers that rely on buss compression as they like the sound. The really
don't need us enlightened engineers telling them what they should and shouldn't
do. This superior attitude among MEs pisses more people off than you can
imagine. Like we're the only ones that know how to implement buss compression.
It's ART Ronny, not some technical paint-by-numbers... Plain and simple Brad,
they are learning how to squash the material too. Squashing is fine, limiting
isn't. Some people like Korn and like the way it sounds. Same with Queens of
the Stone Age, crushed via compression, not by limiting. Sounds kewl. If the
client doesn't want dynamic range, you still have more control adding it in
mastering, rather in mixing and when it's in the digital domain, the less
processing the better, most of the time. Why do you have more control than a
guy with tons of experience and the client right there? It's not brain surgery,
they want their record to sound kewl. You have better gear than TLA or JJP?
They are two different things - that's why they are called different things.
Either way you look at it or what terms we tag on to it, abuse of high
ratio compression is the problem. Agreed. But buss compression is often 2:1
or 4:1 ratios. Nothing wrong with that... Limiting for level = bad. Compression
for sound = good. I'm out...
Brad Blackwood www.euphonicmasters.com

-----------------------------------------------

From: Greg Reierson Date:
Tuesday, June 10, 2003 05:41 PM

Please explain exactly what you can do to a
bass drum when the compression that you add to it, is in the domain of the
other instruments and effecting them as well, that you can't completely do
when the kick is isolated to it's own track? You mean like making the kick
drum pump the vocals. Can't do that in individual tracks. To the contrary, you
have more control of compression on a kick drum, when the compressor is
reacting_only_to it and not to higher level frequencies and peak transients
You have a higher level of individual control of the kick drum, but can not
control the interaction of other mix elements. I'd say you have a 'different'
level of control. One of the basic reasons people use a bus compressor is to
control the MIX as a whole. Same tool, different application. This is the root
of most of our different views. You view limiting and compression as two
different things and I view them as the same process with different settings.
Well, they're both dynamic processes, but they are used so differently that
audio engineers have come up with separate names. It's just more helpful to use
the proper terms, even if the processes themselves seem like two side of a
coin. GR

-----------------------------------------------

From: Ronny Morris
Date: Tuesday, June 10, 2003 07:18 PM

Please explain exactly what you can do
to a bass drum when the compression  that you add to it, is in  the domain
of the other instruments and effecting them as well, that you  can't
completely do when  the kick is isolated to it's own track?   You mean like
making the kick drum pump the vocals. Can't do that in  individual tracks.
You can apply a compressor on the vocal track and have it pump all you want.
When the two buss comped kick is pumping the vocals, it's also pumping the
other instruments. That's fine if that's what you want, but more often pumping
is an undesired effect and introduced by inexperience at the comp settings, at
least on the level of "some" of the mixes that I receive. This doesn't seem to
be a problem on the high profile level, because most of the engineers realize
the brutality of hyper-what-ever-you-want-to-call-it. My arguement is not
against using judicious compression, as Brad puts it in the 2:1 - 4:1 ratios.
He knows this because I've told him this several times. It's the over use of
L1+'s and maximizers on the two buss that lowers the dynamic range, to the
point that further processing is greatly compromised in the mastering stage. An
L1 by any other name is a compressor. So we can nit-pick semantics all day long
and it's not going to change or help the problem. I've heard at least 10
members on this forum in the past month complain about material coming in with
little dynamic range for them to process. Mark my words Greg, it's going to get
worse before it gets better. I'm not trying to dictate how every engineer
should mix, but when I receive material to be mastered that has the life
pancaked out of it and advise the producer to get me a copy with more dynamic
range, the final result is "always" better. Talking my situation and my
clients. What I fail to understand is how we can have a mastering mafia list of
names, dissing ME's for automatically pancaking material, not at the clients
request and not include mix engineers that are doing the exact same thing. The
difference is that "some" mix engineers are doing it and than sending the
material on for further processing. The ME's are doing it at the last stage,
which when used judicioulsy is also fine, but over compression, ok over
limiting, is often detrimental and can not be reversed in this day and time
with any effectiveness, when further processing, such as a mastering eq is
destined. I've had several cases, that In My Humble Opinion, the client got
screwed because the mix engineer left no dynamic range to speak of. If that's
what they want fine, than why seek a mastering suite. I find it hard that
anyone on this forum can disagree about the over use of any processor, when it
takes away from clarity, instrument definition and having normally a dynamic
band sounding like crap. The folks that want you to put a creed level on it,
well they ask for it and I'm like you or anyone else, I'm not going to lose
work over it.  To the contrary, you have more control of compression on a
kick drum, when  the compressor is  reacting_only_to it and not to higher
level frequencies and peak transients   You have a higher level of individual
control of the kick drum, but can not  control the interaction of other mix
elements. I'd say you have a  'different' level of control. One of the basic
reasons people use a bus  compressor is to control the MIX as a whole. Same
tool, different  application.  Agreed and the more the mix engineer applies
processes that are better left for mastering, the more your job is
"comp"romised.

-----------------------------------------------

From: Dave Davis
Date: Tuesday, June 10, 2003 07:43 PM

in article 62491.9367@webbd.nls.net,
Robin Schmidt at r.schmidt@studio-22.de wrote on 6/10/03 7:25 PM:  To make
sure I understand you correctly: Do you mean bus compression over the  whole
mix sum or group busses? Yes. The fact is there are a number of engineers out
there who rely on 2 bus compression for their sound. This is quite deliberate.
In more than one case I know of the intention is to dictate the terms of the
mix to the ME. How loud it will go, how much it will pump etc will all
inevitably be defined by the shape and characteristics of that first layer of
compression. The idea is to reduce things to touch up EQ and overall level
setting in the mastering process. In the specific cases I'm thinking of, he
considers the sound, as delivered, to be the sound. No second guessing allowed.
Period. To be frank, that s not as bad as it sounds, especially when it sounds
good. It makes for a very short day. Sure, there are other cases where things
get out of town, but its important to see that a lot of this is a direct
response to "artistic" mastering, not to mention super aggressive approaches.
You simply can't smash some mixes without destroying them utterly. And that's
the plan.

-----------------------------------------------

From: Bob Katz Date:
Tuesday, June 10, 2003 01:47 PM Dave Davis wrote... In more than one case I
know of the intention is to dictate the terms of the mix to the ME. How loud
it will go, how much it will pump etc will all inevitably be defined by the
shape and characteristics of that first layer of compression. The idea is to
reduce things to touch up EQ and overall Sounds like a vicious circle of
distrust (and loudness) in action. Let's break that circle and get back to good
mixes AND good masters. Producers and engineers and artists go to mastering
engineers to get that neutral perspective the recording usually needs. Start by
sending a first mix to your mastering engineer for his opinion and perspective
on how it's going. A meeting of heads will reveal whether the mix that's coming
out is intentional or perhaps accidental because of innacurate monitoring or
too many late night mix sessions :-). The mastering engineer will respect your
intent, but will also provide that needed perspective. On 6/10/03 1:09:24 PM,
Ronny Morris wrote:  Hypercompression in mastering pop and modern rock is
the norm nowadays, doesn't make sense in teaching them the norm's when there
are better ways to do it. I don't have any objection to them showing the 2-
bus compressor in the course and how to use it, but as Roger Nichols says, "you
should get a license before you use it." Seriously, a day-long lecture-
demonstration, including demonstrating the down side of every devide, will help
the students learn to listen. I object to telling them that they should plug it
in "automatically", and moving on to the "next topic". Recipes... yech.
BK

--------------------------------------------------

From: Bob Katz Date:
Tuesday, June 10, 2003 10:37 PM
How much is too much and how do you know? Has
anyone had the experience of putting on a bonus live cut of the band live made
with simplistic miking, and the live cut blows away all 10 of the
professionally mixed tracks before it? Overuse of bus compression is ONE of the
reasons why this happens more often than you know it.

--------------------------------------------------

From: Glenn Meadows  Is
compressing the 2 bus a band aid for impatient mix engineers? What? You mean
that using the internal compressor on every track to "auto mix" the song isn't
the way it's supposed to be done? A MAJOR engineer who I will NOT mention by
name, does that ALL the time. He adjusts the mix and balance by adjusting the
compressor thresholds, attack, release, virtually NEVER moves the faders, only
uses the automation for mutes. By the time he's done, the faders are all in a
virtual straight line (yardstick mix it used to be called), and the
compressors/limiters on the I/O strips do all the work, along with the eq
tweaks. Then there's the stereo bus compression to flatten it out some more, as
well as slam the analog tape as the last trick, then take that playback and
record it clipping on the digital recorder. Sounds great!!! And all his records
sound the same!!!! glenn

--------------------------------------------------

From: Dave Collins
Date: Wednesday, June 11, 2003 05:06 AM

"If you lift a drum fill at a certain point, you're listening back to the drum fill thinking how
happy you are with it, and then you think 'Oh, but actually the bass disappears
slightly at that point, I think I'll just poke that up,' so you poke the bass
up a bit, and then the third time around you notice that the piano's got
slightly lost at the same point, so you poke the piano up, and you just go
round in circles. So actually what happens is that the mix that had nice highs
and lows, and had some sort of dynamics, is getting slowly flattened out again.
What happens is that the whole mix has just got louder, so you pull the whole
master level down and stick a fat compressor over it, and you've just got a
flat mix, so eight hours later it doesn't sound like you did any work at all.
The amount of people who will tell you that their rough mix was better than the
master mix Ñ it happens all the time. Now why is the rough mix good? It's
probably good because you did it in one hit, you probably said 'Look, just roll
the tape, I'll run off a quick one.' And there'll be things wrong with it, but
if you could just stop there and say 'Look, the only things that are really
wrong are these half-dozen points. There's a bit where the voice gets lost
here, the beginning of that solo which I missed...' And if you could go back
and say 'Let's just take that and tart up those few bits that are wrong using
the computer, so as not to lose the essence of the rough mix,' that should be
your mix. But you never do, because you always think 'This is my chance to make
it even more fabulous!' Gus Dudgeon

--------------------------------------------------

From: Anthony Kitson Date:
Wednesday, June 11, 2003 02:16 AM

I use compressors for SOUND not VOLUME in the
mix. If I want VOLUME I mix it that way! Click here for more discussions on
dynamics in mixing

--------------------------------------------------

John Scrip from Massive mastering wrote:

Some of you are asking why I'm quoting myself... Let's call it a small flurry of interrogative statements that required clarification...

As Bob already explained, the crest is simply the "space between" the average and the peak. More dynamics = larger (wider) crest, less dynamics = smaller (narrower) crest. A mix that has an average level of -20dB(FS)RMS that peaks up around -0dBFS has a crest of 20dB.

What I meant by the above statement was that there are a lot of (usually "less seasoned" for lack of a better term) engineers out there who shoot themselves in the foot by tracking and/or mixing "for volume" -- Destroying the dynamics, using up all the available headroom at the first stage - along with pretty much every subsequent stage.

That's not the way to make recordings that can actually handle "loud" later.

Track with plenty of headroom. Mix with plenty of headroom. Don't throw limiters all over the place just to get the mix "loud" -- Do whatever it takes to make the mix sound *good* and don't be concerned so much with volume.

I'm not saying not to use limiters -- But as a "rule of thumb," if you find a mix actually sounds better - And I mean *BETTER* -- Not "better because it's louder" -- If it actually sounds better being rammed into a limiter, take the limiter off and find out why. Maybe one thing in that mix is truly "too dynamic" for the rest of the mix. Put the limiter on THAT and see how it sounds. Use a compressor when something has a dynamic range that's too wide for the mix -- Not because someone told you that everybody compresses everything so it can be louder.

"Punch" and "impact" comes from the difference between loud and quiet - Not the absence of quiet. And although I'm not a fan of the current "level insanity" going on, it's better to have decent sounding recordings that have the potential to be loud than loud sounding recordings that should be shut off. Almost invariably, it's those dynamic, wide-crest mixes that have that potential.

mixing "hot" doesn't do anything to make your finished product louder. Tracking hot doesn't either (I don't even want to get into the nastiness that can happen from tracking too hot). Headroom is good room. Keep it, love it, cherish it - Your mastering guy (even if that's yourself, which I also won't get into) is almost undoubtedly going to use it all up... Give him some room to move.

 

More Discussions from Recording Engineers on the Mastering Webboard

Topic: major dynamic swings on pre-master
From: Marshall Simmons Date: Thursday, July 31, 2003 05:14 AM

Hello--I'm about to send out some mixes to be mastered on a project that has quite a bit of promise... I've recorded everything with very little compression to the daw, and i'm finding that as i'm finishing up the mixes, i realize that i'm not using very much compression in the daw. It sounds pretty small at a 75db refrence
level, but bring it up to 95db refrence level and its HUGE. Dynamic impact and lots of space around all the instruments. Just so you know, this is more of a Emo--somewhat space-rock style of music. The difference between peaks and RMS right now is about 15 db. My question is this: Is this ok? I know we have all talked about the lack of dynamic range in recordings nowadays, and thats what i want to avoid, but i'm afraid i'm leaving the mastering engineer too much dynamic range. How i mixed is that i set levels without any compression or eq, listened, and then looked for instruments that poked out to much on transients, or got buried in soft parts... I just used enough compression on those instruments orvocals to keep them from poking out too much or not dissapearing without changing my overall balance. No compressor or limiter on the 2buss.. I guess its late and i'm rambling...I just got back from the session and this is the first group that i've done that has a chance of going somewhere...Thoughts and opinions would be greatly appreciated.

Marsh
________________________________________________________________
From: Andreas Balaskas
Date: Thursday, July 31, 2003 05:59 AM

Marshall, there's never too much dynamic range to have (we can always squash it to death ;-)) Sounds as one of us is going to have fun mastering it! masterlab mastering studios

andreas balaskas
________________________________________________________________
From: Ronny Morris
Date: Thursday, July 31, 2003 07:29 AM

Emo must sound something Eno than.
> My question is this: Is this ok? I know we have all talked about the lack
> of dynamic range in recordings nowadays, and thats what i want to avoid,
> but i'm afraid i'm leaving the mastering engineer too much dynamic range.
I see some rock with -20dB RMS average and orchestra around -25dB or more once in awhile. Lot's of music at -15dB. There really isn't a standard on using RMS for perceived levels, as all music is different from one song to the next. Things like dead air breaks, fade ins/outs, long decay sections etc. can give you a lower RMS reading on a song that sounds louder, than one with higher RMS.A fade in from black to -3dB, that occurs in a fast crescendo, will sound louder than a gradual rise in gain from -15 to -3dB. Even though the peaks are thesame and the RMS average is louder on the -15dB to -3dB section.
> How i mixed is that i set levels without any compression or eq, listened,
> and then looked for instruments that poked out to much on transients, or
> got buried in soft parts... I just used enough compression on those
> instruments or vocals to keep them from poking out too much or not
> dissapearing without changing my overall balance.
> No compressor or limiter on the 2buss...
Sounds like you've got it sounding good to you. That's what counts and speaking for myself, it's by far better to leave "too much dynamic range" in, when going to mastering than not leaving enough. Many will agree that most of the problems come from not having enough dynamic range, these days and wish more people
mixed at your dynamic level. I'm not sure if there is really a thing called "too much dynamic range", unless you are trying to compete in a genre market and IMHO, that falls under perceived leveling and is best left at the mastering stage.

Ronny Morris RoMo Recording Studios Brunswick, Ga. USA
________________________________________________________________
From: Robin Schmidt
Date: Thursday, July 31, 2003 07:48 AM

I agree with all of the above. Assuming that you can pass on high resolution material, you can never have enough dynamic range in the mix. I've had source files come in that had a measured RMS of less than -30 dB (well, on jazz and classical, that is) and there is really no reason against it. An RMS level of -15db is absolutely OK, even if you can only pass on a 16 bit mix.

Robin Schmidt 24-96 Mastering
________________________________________________________________
From: Bob Olhsson
Date: Thursday, July 31, 2003 12:55 PM

On 7/31/03 5:14:11 AM, Marshall Simmons wrote:
>How i mixed is that i set levels without any compression or eq, listened, and then
>looked for instruments that poked out to much on transients, or got buried in soft parts...
>I just used enough compression on those instruments or vocals tokeep them from
>poking out too much or not dissapearing without changing myoverall balance.
The next step is riding the gain in those spots rather than using a compressor. Of course then it'll sound "vintage" in addition to sounding huge! It's too bad more people aren't taking the time to mix the way you just did. In addition to sounding huge, it'll also
sound the same almost anywhere it gets played.
________________________________________________________________
From: Paul Gold
Date: Thursday, July 31, 2003 10:01 PM

On 7/31/03 12:55:00 PM, Bob Olhsson wrote:
>The next step is riding the gain in those spots rather than using a compressor.
My, my. Spending the time to do it right. What will they think of next. That's one reason I enjoy live mixing of large and/or acoustic groups. I always ride faders and almost never use compressors. It can sound huge with a powerful high headroom system.

Paul Gold
________________________________________________________________
From: Marshall Simmons
Date: Friday, August 01, 2003 02:19 AM

Thank you all for you wonderful comments... I will definitelytry the hand riding of the levels instead of compressing. So far the only thing really needing compression is the bass and the vocals. I've been recording at 24b/44.1k and trying to use minimal and distant miking for almost everything. Drum overheads were a pair of schoeps omnis and and audix d4 for kick. Distorted and clean
guitars were through a AEA r84 ribbon mic and a shure ksm32 ontwo different amps (panned l and r--double tracked) spacey guitar was with a AB schoeps omni setup about 5 feet away from the stereo amp array acoustic guitar was with the R84 about 6 feet back and the vocals were through the R84 preamps were ward beck 460A and a benchmark 400 preamp. I monitor in the farfield with
Paradigm studio reference 40's (i love those speakers) I don't have alot of equipment and the DAW definitely leaves something to be desired (M audio delta 66 into nuendo) but i figured that i'd try to keep it simple all the way through to get the best sound i could. I never liked nearfieldmonitoring, and the audio fool in me is used to listening in the far field, so thats what i do when i mix Sorry to ramble again. If yall are interested, when I'm finished I'll try to
find a place to upload some examples. I could definitely use the advice.

Thanks again Marsh
________________________________________________________________
Conf: GEAR-WHAT'S HOT!/WHAT'S NOT!
From: Robin Schmidt
Date: Friday, August 01, 2003 06:03 AM

And I'd definitely like to hear it ... interesting miking... I didn't know Schoeps were used much in rock music. Please do let me know when it's finished.

Robin Schmidt 24-96 Mastering
________________________________________________________________

Topic: Monitor Madness!!!!
Conf: GEAR-WHAT'S HOT!/WHAT'S NOT!
From: Aron Gillman
Date: Monday, August 04, 2003 09:24 AM

Now, I do not expect anyone to tell me how to tighten up the bass or how to deal with a compressed stereo mix. But I do have one incredibly hard question which is very daunting... On a scale of one to ten, how much can one fix an unbalanced mix where there is no kick, or the vocal is so sibilant than de-essing ends up removing all the highs along with the top of the snare. When the kick and snare peak 6db with every hit and the low end pumps the woofers before I even touch anything. Will that mix always sound like that mix, only more balanced? It seems that even when I try to use a technique like Bob Katz's for setting up the compressor to react to those instruments or ranges, it still sounds very similar to me. Don't get me wrong there is a big overall improvement in the smoothness, but they still sound like they control the dynamic range. It just leaves me feeling like I could have done some thing to fix that.( Do remember I approach a lot of what I do with some of my idiosyncrasies of my mixing background, I'm trying to let some of them go) Is the mix what it is at it's root??? I only ask
because I do believe that the majority of those who may read this post could probably make some sense of this with a number between 1-10. I knowit most likely depends on whose working on the mastering, but it would very much help me set a standard for myself and what I can expect to achieve. I hope that all makes sense in a way. I feel pretty bad lately for all the mastering
engineers who had to work on some of my mixes before I gained some knowledge as to what I was doing behind a board... I used to compressmy stereo buss like a brick wall... Karma huh???

Thanx again Aron
________________________________________________________________

Topic: that monitor thing...
Conf: GEAR-WHAT'S HOT!/WHAT'S NOT!
From: Bob Katz Date: Monday, August 04, 2003 10:32 AM

Hi, Aron. The art of it is knowing when a mix is so bad that it really can't be "fixed," and that anything you do would be a bandaid or make other things much worse, exactly as you have described! I know it's impossible to describe how to know
in words, and only experience will tell, but I will say that in myyears, if a recording does not sound pretty good to begin with, then any tomfoolery you try will probably result in a tomfooled master. I said "probably" because improbably, once in a while, using tricks such as M/S splits and multiband dynamics processing, it has been possible to turn a sow's ear into a respectable purse. But don't expect those kind of magic tricks to work and waste your time over it except as an exercise to learn what you can't do. We may try to do this, for example, in the context of a "historic" recording that has to be included on an album. Yet there are some
amazing "miracles" you can accomplish. Bass problems, such as a bass sound that is too fat and thick and unclear, can often be improved without hurting the other instruments too much, using M/S and other techniques. A vocalthat is just a little bit under can be brought out, and vice versa. But the overall recording has to have some sonic merit underneath its defects, or it is probably fruitless to proceed.

Bob Katz
________________________________________________________________
Topic: Monitor Madness!!!!
Conf: GEAR-WHAT'S HOT!/WHAT'S NOT!
From: Lynn Fuston Date: Monday, August 04, 2003 11:27 AM

On 8/4/03 9:24:00 AM, Aron Gillman wrote:
>On a scale of one to ten, how much can one fix an unbalanced mix where there.........
I have always maintained that a well-balanced mix I can take anywhere. You can do just about anything with it. For instance, I just had mix come in that had substantial low end (80 Hz on down), a big bump in the low mids (3 dB too much in the 175-250
region) and was enormously soft on top (probably about a 6 dB deficiency from 8K on up). But the balance, the mix, was great. I spoke at length with the new client, telling him he obviously was very talented but I thought these universal characteristics of all his mixes seemed to indicate that there were severe issues with his monitoring. I was able to fix the problems and the mixes held up very well. So a well balanced mix with issues is far desirable to a good mix with fatal-flaw problems. The bass/kick ratio is one of those, though that is occasionally remediable with multiband compression. The "no music" syndrome is another, where the rhythm and vocals are hitting hard but the mids, all the music-the keys, guitars, strings- are sucked out. That one is a non starter.

Lynn Fuston 3D Audio Inc
music mastering and mixing
On a scenic hilltop outside of historic Franklin, Tennessee
________________________________________________________________

Also see Mixing Tips and Tricks by Bob Katz

Communication Science

A story came to me, and it reminds me that communication still is a global challenge. You might think that your message is clear, but an ever so fine deviation, can completely change the meaning of your message.

"The Gift"

A young man wanted to purchase a Christmas gift for his new sweetheart, and as they had not been dating very long, after careful consideration, he decided a pair of gloves would strike the right note: romantic, but not too personal. Accompanied by his sweetheart's younger sister, he went to Harrods and bought a pair of white gloves. The sister at the same time purchased a pair of panties for herself. During the wrapping, the clerk mixed up the items and the sister got the gloves and the young man got the panties.

Without checking the contents, the young man sealed the package and sent it to his sweetheart with the following note: "I chose these because I noticed that you are not in the habit of wearing any when we go out in the evening. If it had not been for your sister, I would have chosen the long ones with the buttons, but she wears short ones that are easier to remove. These are a delicate shade, but the lady I bought them from showed me the pair she had been wearing for the past three weeks and they were hardly soiled. I had her try yours on for me and she looked really smart. I wish I was there to put them on for you the first time, as no doubt other hands will come in contact with them before I have a chance to see you again. When you take them off, remember to blow in them before putting them away as they will naturally be a little damp from wearing. Just think how many times I will kiss them during the coming year. I hope you will wear them for me on Friday night. All my love. Barry P.S. The latest style is to wear them folded down with a little fur showing.

---

Penalties for Bass Infractions

NAME OF OFFENDER (Bass Player)_______________________
INFRACTION DATE___________________________

MUSICAL OFFENSES
Playing loudly during warm up $10
Sound-checking amp with funk slapping $25
Loud cursing after mistake $10
Playing high and fast after mistake $20
Practicing 2-handed tapping between tunes $20
Asking for "E" tuning note $25
Playing E anyway when horns tune to Bb $50
Playing written-out walking line $50
Failure to play written walking line $75
Writing note names over ledger-line notes $50
Writing beat numbers under dotted figures $50
Playing eighth notes $5 each
Playing sixteenth notes $10 each
Playing above 1st octave immediate dismissal
Dragging fast tempo $75
Dragging ballad tempo $100
Blacking out during ballad $200
Ignoring drummer's tempo $100
Following drummer's tempo $250
Asking to borrow Real Book for All Of Me $1000

UPRIGHT PLAYERS
Showing up before first downbeat $25
Playing audibly $25
Faking changes $25
Slapping $150
Missing tutti lick, then mentioning vintage of bass $25
Excessive sweating $25
Pedal point double-stops during horn solo $50
Asking leader for a solo $30
Accepting solo when offered $50
Taking second chorus $100
Playing solo arco $400
Pretending to check tuning after playing out of tune $100
Playing "A Train" ending on every tune $200
Playing extended "A Train" ending on every tune $500

ELECTRIC PLAYERS
Checking hair between tunes $15
Experimenting with odd meters $25
Missing root at end of blistering fill $25
Playing with a pick $50
Tuning during ballad $30
Playing Jaco groove on samba $75
Playing Jaco samba groove on ballad $150
Attempting last word on final chord $50
Achieving last word on final chord $100
Long gliss down to final note $200

EQUIPMENT VIOLATIONS - ELECTRIC
Forgetting strap $10
Changing strings after every set $15
Using electric tuner $15
Setting up mic "just in case" $75
Forgetting to turn amp on $40
Bringing amp larger than 1 person can carry in 1 trip $50
Asking horn player for help moving amp $25
Bringing custom-made bass $100 per string above 4
Bringing more than 1 bass $100 per extra bass
Skull decals on bass $150
Bringing fretless bass $500

CRIMINAL BAD TASTE
Telling bone player about all the gigs you get $10
Asking bone player about their day gig $10
Sitting behind drums on break $10
Quoting "Birdland" $25
Practicing scales during break $25
Practicing scales during drum solo $50
Practicing $150
Beginning a sentence with "When I was a guitar player..." $50
Casually mentioning to Musical Director of cheap theater that you are "into sequencing" $100

BASIC STUPIDITY
Wearing old Buddy Rich tour shirt $10
Wearing new Whitesnake tour shirt $20
Asking when the rock set starts $20
Continually asking "where are we?" $25
Continually shouting "Yeah!" $25
Asking bone player where "1" is $50
Taking cell phone call during 4's $100

---

The New Pope's Ritual with the Chief Rabbi

The Ritual

Every time a new pope is elected, there's a whole lot of rituals and ceremonies that have to be gone through, in accordance with tradition. Well, there's one tradition that very few people know about. Shortly after the new pope is enthroned, the chief rabbi seeks an audience. He isshown into the pope's presence, where upon he presents him with a silvertray bearing a velvet cushion. On top of the cushion is an ancient, shriveled parchment envelope. The pope symbolically stretches out his arm in a gesture of rejection. The chief rabbi then retires, taking the envelope with him and does not return until the next pope is elected.

John Paul II was intrigued with this ritual, whose origins were unknown to him. He instructed the best scholars of the Vatican to research it, but they came up with nothing. When the time came and the chief rabbi was shown into his presence, he faithfully enacted the ritual rejection. But, as the chief rabbi turned to leave, he called him back. "My brother," the holy father whispered, "I must confess that we Catholics are ignorant of the meaning of this ritual enacted for centuries between us and you, the representative of the Jewish people. I have to ask you, what is it all about?" The chief rabbi shrugs and replies: "but we have no more idea than you do. The origin of the ceremony is lost in the mists of ancient history." The pope said: "let us retire to my private chambers and enjoy a glass of wine together, then, with your agreement, we shall open the envelope and discover at last the secret." The chief rabbi agreed.

Fortified in their resolve by the wine, they gingerly pried open the curling parchment envelope and with trembling fingers, the chief rabbi reached inside and extracted a folded sheet of similarly ancient paper. As the pope peered over his shoulder, he slowly opened it. They both gasped with shock. It was the check for the last supper.

---

Industry Sharks

A True Story from Alan Kefauver. How he avoids industry sharks who do telephone solicitation.

Part of my son's allowance is based on his job as phone answerer after 5:00. He loves to chat up the salesmen. He's 8. He tries to act like an adult, appears interested, then when they try to close the sale he says, "I don't have any money, I'm 8." After realizing they have wasted time talking to an 8 year old, they never call back.

Cheers

Alan P. Kefauver, Director
Recording Arts and Sciences
Peabody Institute of The Johns Hopkins University
http://www.peabody.jhu.edu/recording-arts

---

Market Research

A market researcher called at a house and his knock was answered by a young woman with three small children running around her. He asked her if she minded replying to his questions and when she agreed, he asked her if she knew his company, Cheeseborough-Ponds. When she said no, he mentioned that among their many products was Vaseline and she certainly knew of that product. When asked if she used it, the answer was "yes". Asked how she used it, she said "to assist sexual intercourse."

The interviewer was amazed. He said, "I always ask that question because everyone uses our product and they always say they use it for the child's bicycle chain, or the gatehinge; but I know that most use it for sexual intercourse. Since you'vebeen so frank, could you tell me exactly how you use it?" "Yes, we put iton the doorknob to keep the kids out."

---

Lawyer's Charity

Lawyer #1: The local United Way office realized that it had never received a donation from the town's most successful lawyer. A local volunteer calls to solicit his donation, saying, "Our research shows that even though your annual income is over a million dollars, you do not give one penny to charity! Wouldn't you like to give back to your community through The United Way?" The lawyer thinks for a moment and says: "First, did your research show that my mother is dying after a long, painful illness and has huge medical bills far beyond her ability to pay?" Embarrassed, the United Way rep mumbles, "Uh, no." "Secondly, that my brother, a disabled veteran, is blind and confined to a wheelchair and is unable to support his wife and six children?" The stricken United Way rep begins to stammer an apology but is cut off. "Thirdly, that my sister's husband died in a dreadful traffic accident," the lawyersvoice rising in indignation, "leaving her penniless with a mortgage andthree children?" The humiliated United Way rep, completely beaten, says simply, "I had no idea." The lawyer then says, "Well then ... and ifI don't give any money to THEM, why should I give any to you?"

Lawyer #2: "The Dying Man's Wish"
A very wealthy man, old and desperately ill, summons to his bedside his three closest advisors: his doctor, his priest, and his lawyer. "I know," he says, "they say you can't take it with you. But who knows? Suppose they're mistaken. I'd like to have something with me, just in case. So I am giving each of you an envelope containing one hundred thousand dollars and I would be grateful if at my funeral you would put the envelopes in my coffin, so that if it turns out that it's useful, I'll have something. They each agree to carry out his wish.Sure enough, after just a few weeks, the old man passes away. At his funeral, each of the three advisors is seen slipping something into the coffin.

After the burial, as the three are walking away together, the doctor turns to the other two and says, "Friends, I have a confession to make. As you know, at the hospital we are desperate because of the cutbacks in funding. Our CAT SCAN machine broke down and we haven't been able to get a new one. So, I took $20,000 of our friend's money for a new CAT SCAN and put the rest in the coffin as he asked."At this the priest says, "I, too, have a confession to make. As you know, our church is simply overwhelmed by the problem of the homeless. The needs keep increasing and we have nowhere to turn. So I took $50,000 from the envelope for our homeless fund and put the rest in the coffin as our friend requested."Fixing the other two in his gaze, the lawyer says, "I am astonished and deeply disappointed that you would treat so casually our solemn undertaking to our friend. I want you to know that I placed in his coffin my personal check for the full one hundred thousand dollars."

---

Translation Guide to a Recording

Session Musician to engineer. "Could we have more band in the phones?"
Translation: "The singer is too f**king loud in the phones !

"Singer to engineer: "I can't hear myself.
"Translation. "I don't want to hear anyone but myself.

"Musician to guitarist: "Can you hear yourself okay?
"Translation: "You're too f**king loud in the phones !"

Bassist to band. "Can everybody hear the drums?"
Translation: "This band is swinging like a broken record !

"Drummer to bassist: "Can you hear the kick drum?"
Translation: "We're not locking..... !!

Bassist to producer: "Could we have more steel/fiddle/accordion in the phones?"
Translation: "I will punish the band for rushing."

Musician to producer: "Could we have more piano in the phones?"
Translation: "Your artist can't sing in tune.

"Musician to writer: "This song has nice changes.
"Translation. "It's amazing what you can do with two chords.

"Musician to producer or artist: "This song sounds like a hit.
Translation: "This song sounds like another song."

Producer to band: "It's a feel thing."
Translation: "I know the song sucks, but the artist wroteit."

Musician to producer: "I don't think we'll beat the magic of that first take."
Translation: "Please don't make us play this piece of s**t again."

Drummer to band: "Should we speed up the tempo a couple of clicks?"
Translation: "Do you all intend to keep rushing?

Musician: "Could we listen to one in the control room?"
Translation: "The way these phones sound, we might as well be listening to Radio Free Europe."

Producer to band: "Let's take a break and come back and try one more."
Translation: "I think I'm having a nervous breakdown."

Musician to producer- "Were we booked for two sessions today?"
Translation: "Another three hours of this and I may have to kill you."

Producer to band: "We're supposed to be done at six, but we've got only one more tune and I was wondering if we could skip our dinner break and work straight through."
Translation: "You'll be done at nine, and you'll be hungry."

Artist to producer: "I don't like this song; it really sucks.
"Translation: "I didn't write this song."

Producer to artist: "Trust me. It is a good song. Radio will love it."
Translation: "F**k you, I own the publishing on this song.

Singer to musician: "Can you play something like (so-and-so) would play?"
Translation: "I really wanted (so-and-so) on this record."

---

I really wish this happened...

One night, a police officer was staking out a particularly rowdy bar for possible violations of driving-under-the-influence laws. Watching from his squad car, he saw a fellow stumble out the door, trip on the curb and try 15 cars before finding his own and promptly falling asleep in the frontseat. As the evening progressed, the owners of other cars left the bar and drove away. Finally, the sleeper awakened, started his engine and began to pull away. The police officer pounced, waved him to a halt and administered a Breathalyzer test. The results showed a 0.0 blood-alcohol level. The puzzled officer demanded to know how that could be.

The driver replied, "Tonight, I'm the designated decoy."

---

Who Designed the Human Body?

Three engineering students were gathered together discussing the possible designers of the human body. One said, ``It was a mechanical engineer. Just look at all the joints.'' Another said, ``No, it was an electrical engineer. The nervous system has many thousands of electrical connections.'' The last said, ``Actually it was a civil engineer. Who else would run a toxic waste pipeline through a recreational area?''

---

A Priest, A Lawyer, and an Engineer

In some foreign country a priest, a lawyer and an engineer are about to be guillotined. The priest puts his head on the block, they pull the rope and nothing happens -- he declares that he's been saved by divine intervention -- so he's let go. The lawyer is put on the block, and again the rope doesn't release the blade, he claims he can't be executed twice for the same crime; he is set free, too. They grab the engineer and shove his head into the guillotine, he looks up at the release mechanism and says, "Wait a minute, I see your problem......"

---

America Offline, Inc.

Free Software Helps You Get Offline in 5 Easy Steps! Every day you hear more and more people talking about it. You hear about it at work, in restaurants, even on the bus. More and more Americans are discovering how much more time theyhave, how much easier it is to manage their money, and personal andbusiness relationships when they do it offline. Our free software includes utilities to seek outand destroy all remnants of communications software on your computer (also works over a network). It even removes that peskylittle winsock.dll from your hard drive once and for all! Saygood-bye to theWorld Wide Web and hello to the friends and familyyou forgot you had! Our software, once run, remains memory-resident,and like a background virus checker, protects your computerfrom re-installation of telecommunications software. Here's how the program works: 1. First you run the good-bye letter generation program which automatically logs you into all your favorite online services and posts good-bye notes to the news groups and message areas of your choice. It encourages people to call or even writeyou if they want to talk to you, and lets them know that you'regoing to be okay, but that you're just going offline. (It even logs you into your favorite chat areas and makes witty parting comments, and leaves requests with systems administrators to cancel your account.) 2. Our program recognizes every version ofevery known communications software package for DOS, Windows,Windows 95, Windows NT, OS/2 and Unix, (and there's even a special version for you Mac Users.) It will remove all of these programs andoverwrite your hard drive with null strings so they can't beundeleted. And just so you don't try to sneak in through a back doorlike an automated checkpayment window through a program likeQuicken, our software also removes the serial communications abilities of all financial software and related software. 3. Hardware reconfiguration. Our software alters your system configuration so that no device which even remotely resembles a modem will ever work on your computer again. It installs a "listening program" as a permanent TST so that even if your computer encounters a modem tone over a network, your computer will immediately reboot, thereby keeping you effectively offline, even in a network environment. 4. Hardware destruction. The next step is to destroy your modems. This is best done with a hammer. We recommenda hammer because we know you will start to feelin that destructive release the first real joy of your new life offline.With each crushing blow, we encourage you to think of the countless hours you've wasted, the completely irrelevant information you have gathered, and the many people who completely mis-represented themselves to you when you were online. 5. Finally, our software brings you a brief lesson on meditation to help you begin to relax without being online. It also includes a list of things that you can do offline, like feeding your cat that you forgot you had, watering the plants, doing the dishes, seeing your friends in person, writing notes on pieces of paper, calling your parents on their birthday insteadof sending e-mail. The list includes more than 1,000 offlineliving tips. We're sure you'll agree that this software isthe best of its kind for getting offline quickly and staying this way. To get your free software, please send a hand-written note to America Offline, Inc., 4578 Chestnut Drive, Emporia, KA 83903. And please include a stamped, self-addressed envelope. Please specify your operating system and 3.5 or 5.25 inch disks. Note: Our software is not available on CD ROM, we do not have a support BBS or a Web page or even a fax machine. OUR SOFTWARE IS NOT AVAILABLE ONLINE. UPLOADING IT IS EXPRESSLY FORBIDDEN! Top Ten Reasons: "Men Prefer Computers to Women" 10.) Computers have "help" text when you get confused. 9.) When you get tired of a computer, you can shut it off. 8.) Your friends will always tell you, you could have abetter one. 7.) Booting is not a punishable offense. 6.) You can get upgrades without going to a plastic surgeon. 5.) "Cheap" and "Fast" are good attributes in a computer. 4.) Nobody stares if you finger a computer in your office. 3.) A computer doesn't get mad if you play with someoneelse's computer. 2.) It only takes a couple of seconds to turn one on. AND THE NUMBER 1 REASON.......... 1.) Computers will take a 3-1/2 floppy & be happy about it. WHY DID THE CHICKEN CROSS THE ROAD? GEORGE W. BUSH: We don't really care why the chicken crossed the road. We just want to know if the chicken is on our side of the road or not. The chicken is either with us or it is against us. There is no middle ground here. AL GORE: I invented the chicken. I invented the road. Therefore, the chicken crossing the road represented the application of these two different functions of government in a new, reinvented way designed to bring greater services tothe American people. RALPH NADER: The chicken's habitat on the original side of the road had been polluted by unchecked industrialist greed. The chicken did not reach the unspoiledhabitat on the other side of the road because it was crushed by the wheels of a gas-guzzling SUV. PAT BUCHANAN: To steal a job from a decent, hard-working American. RUSH LIMBAUGH: I don't know why the chicken crossed the road, but I'll bet it was getting a government grant to cross the road, and I'll bet someone out there is already forming a support group to help chickens with crossing-the-road syndrome. Can you believe this? How much more of this can real Americans take? Chickens crossing the road paid for by their tax dollars, and when I say tax dollars,I'm talking about your money, money the government took from you to buildroads for chickens to cross. JERRY FALWELL: Because the chicken was gay! Isn't it obvious? Can't you people see the plaintruth in front of your face? The chicken was going to the "other side." That's what they call it -- the other side. Yes, my friends, that chickenis gay. And, if you eat that chicken, you will become gay too. I say we boycottall chickens until we sort out this abomination that the liberal media whitewasheswith seemingly harmless phrases like "the other side." DR. SEUSS: Did the chicken cross the road? Did he cross it with a toad? Yes, the chickencrossed the road, but why it crossed, I've not been told! ERNEST HEMINGWAY: To die. In the rain. Alone. MARTIN LUTHER KING, JR.: I envision a world where all chickens will be free to cross roads withouthaving their motives called into question. GRANDPA: In my day, we didn't ask why the chicken crossed the road. Someone told usthat the chicken crossed the road, and that was good enough for us. BARBARA WALTERS: Isn't that interesting? In a few moments we will be listening to the chickentell, for the first time, the heart-warming story of how it experienced aserious case of molting and went on to accomplish its life-long dream ofcrossing the road. JOHN LENNON: Imagine all the chickens crossing roads in peace. ARISTOTLE: It is the nature of chickens to cross the road. KARL MARX: It was a historical inevitability. SADDAM HUSSEIN: This was an unprovoked act of rebellion and we were quite justified in dropping50 tons of nerve gas on it. VOLTAIRE: I may not agree with what the chicken did, but I will defend to the deathits right to do it. RONALD REAGAN: What chicken? CAPTAIN KIRK: To boldly go where no chicken has gone before. FOX MULDER: You saw it cross the road with your own eyes! How many more chickens have to cross before you believe it? SIGMUND FREUD: The fact that you are at all concerned that the chicken crossed the roadreveals your underlying sexual insecurity. BILL GATES: I have just released eChicken 2003, which will not only cross roads, butalso will lay eggs, file your important documents, and balance your checkbook - and Internet Explorer is an inextricable part of eChicken. MARTHA STEWART: No one called to warn me which way that chicken was going. I had a standingorder at the farmer's market to sell my eggs when the price dropped to acertain level. No little bird gave me any insider information. ALBERT EINSTEIN: Did the chicken really cross the road or did the road move beneath the chicken? BILL CLINTON: I did not cross the road with THAT chicken. What do you mean by chicken?Could you define chicken, please? THE BIBLE: And God came down from the heavens, and He said unto the chicken, "Thou shaltcross the road." And the chicken crossed the road, and there was much rejoicing. COLONEL SANDERS: I missed one?

THANK YOU FOR CALLING THE MENTAL HEALTH HOT LINE

If you are obsessive/compulsive, please press 1 repeatedly.
If you are codependent, ask someone to press 2 for you.
If you are a multiple personality, press 3, 4, 5, 6.
If you are paranoid and delusional, we already know what you want and who you are. Just stay on the line while we trace your call.
If you are schizophrenic please hold, and a little voice will tell you what to press.
If you are depressed it probably doesn't matter what you press, it won't do you any good anyway...

---

Two Writers and an Editor

Two writers and an editor were dragging themselves across an endless desert, when suddently they spied a green oasis. Mirage? No! As soon as they reached it, the writers stripped off their clothes and started frolicking in the pond. Suddenly, they noticed the editor standing on the bank, peeing into the water.

"What are you DOING!?" they asked. And he replied: "I'm making it better."

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This one from Ivan Berger's collection (now you know who to blame)

I just read that the Pentagon purchased washers the size of nickels from Lockheed for $230 each... Coincidentally, it's tax time, and I seem to owe quite a bit of money... Enclosed please find six lugnuts. You can keep the change.

Sincerely,

Z.

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Heaven and Hell

Do you know the difference between heaven and hell?

In heaven all the cooks are French, the mechanics are German, all the lovers are Italian, and everything is run by the Swiss.

In hell, all the mechanics are French, all the lovers are Swiss, the cooks are English, and everything is run by theItalians.

(I don't know the source of this one. Hope it hasn't offended anyone, either. If anyone has something to add about Americans (United States), then we'll cookup an addition to this joke)

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PC versus Mac

From Rec.Humor.Funny 2/22/95
Sorry I lost the writer of this cute one...

Q: Why is it that Mac can read DOS disks, run DOS software,etc., but IBM machines can't handle Mac stuff?

A: Well it's normal for a system to be downwardly compatible.

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Mabel and Ethel

Mabel and Ethel were walking down the street one day, when Ethel spied a frog sitting on the sidewalk. "Helpme, help me!" the frog said, "I used to be a jazz saxaphonist, butan evil witch turned me into a frog. The only way to break the spellis to kiss me. Help me, help me!"

So Ethel picked the frog up, put it in her pocketbook, and they continued to walk along.

Shortly, the frog popped out of Ethel's pocketbook,and repeated (in its inimitable way), "Help me, help me! I usedto be a jazz saxaphonist, but an evil witch turned me into a frog.The only way to break the spell is to kiss me. Help me, help me!"

Ethel picked the frog up, put it back in her pocketbook, and they continued to walk along. "Why'd you do that?" asked Mabel. "Why didn't you kiss the frog and break the evil spell?" "Frankly," replied Ethel, "you can make a hell of a lot more money with a talking frog than with a jazz saxaphonist!"

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Love is Blind...

A blind rabbit met a blind snake in theforest. The rabbit told the snake, "I don't know what I am. Couldyou please feel me and tell me what I am?" So the snake felt the rabbit,"You've got soft fur, a cottony tail, and big ears...you must be arabbit. Now could you do the same for me? I don't know what I am, either."

The rabbit felt the snake. "You've got slimy skin, a long tail, and no ears. You must be a record producer!"

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Bill Clinton, Boris Yeltsin, and Bill Gates

Bill Clinton, Boris Yeltsin, and BillGates were called in by God. God informed them that he was very unhappy about what was going on in this world. Since things were sobad he told the 3 that he was destroying the Earth in 3 days. They were all allowed to return to their homes and businesses and tell their friends and colleagues what was happening. God did tell them though that no matter whatthey did he was *not* changing his mind. So...

Bill Clinton went in and told his staff... I have good news and badnews for you. First the good news.... there *is* a God. The bad news is that he is destroying the Earth in 3 days.

Boris went back and told his staff that he had BAD news and more BAD news. The first was... there *is* a God... the second was that he was destroying the Earth in 3 days.

Bill Gates went back and told his staff... I haveGOOD news and more GOOD news... first... God thinks I am one of the 3 most important people in the world. The second is... we don't have to fix the bugs in Windows95!

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Music Related One-Liners

Why did the producer cross the road?
Cuz that's the way the Beatles did it, man.

How many producers does it take to change a light bulb?
I dunno... what do *you* think?

What's the difference between a dressmaker and a viola player?
A dressmaker tucks up frills.

How do you know there's a drummer at your door?
The knocking speeds up.

How do you know there's a singer at your door?
He doesn't know when to come in.

How many guitarists does it take to change a light bulb?
One to do it and ten to stand around and say "I cando that".

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A Hundred Computer Related One-Liners

WARNING: No user serviceable characters in these taglines.

------------------------------------------------------------------------

Backups? We doan *NEED* no steenking baX%^~,VbKx NOCARRIER

Is reading in the bathroom considered Multi-Tasking?

My computer isn't that nervous...it's just a bit ANSI.

If only women came with pull-down menus and online help.

My computer's sick. I think my modem is a carrier.

Gotta run, the cat's caught in the printer.

Honey, I Formatted the Kid!

Spelling checkers at maximum! Fire!

Your e-mail has been returned due to insufficient voltage.

Who is General Failure and why is he reading my disk?

Hex dump: Where witches put used curses...

Finish your mail packet! Children are offline in India.

Never violate the Prime Directory! C:

Multitasking: Screwing up several things at once...

Maniac: An early computer built by nuts...

Stack Error: Lost on a cluttered desk...

Stack Overflow: Too many pancakes...

Terminal glare: A look that kills...

Trojan: Storage device for replicating codes...

ZMODEM: Big bits, Soft blocks, Tighter ASCII...

Life would be much easier if I had the source code.

Mommy! The cursor's winking at me!

Managing programmers is like herding cats.

Do files get embarrassed when they get unzipped?

Capt'n! The spellchecker kinna take this abuse!

C:BELFRY is where I keep my .BAT files.

ASCII to ASCII, DOS to DOS.

"Mr. Worf, scan that ship." "Aye, Captain... 300 DPI?

How do I set my laser printer on stun?

The best way to accelerate a Mac is at 9.8 m / sec^2

"!sgub evah t'nseod CP sihT ?sgub naem ayaddahW"

"E=Mc^5...nahhh...E=Mc^4...nahh...E=Mc^3...ah, the hell with it."

"Today's subliminal thought is:"

'Profanity: the universal programming language'

'Calm down -- it's only ones and zeros.'

'.... now touch these wires to your tongue!'

Computer analyst to programmer: "You start coding. I'll go find out what
they want."

LSD: virtual reality without the expensive hardware.

According to my calculations the problem doesn't exist.

C:GRAPHICSGIFNAUGHTYFILTHYDISGUSTINGWOW!

Computer Science: solving today's problems tomorrow.

It said, "Insert disk #3," but only two will fit!

RAM DISK is not an installation procedure!

Computers are only human.

Was that your wife I saw in that GIF?

I used to have a life, then I got v32bis!

If the pen is mightier than the sword, and a picture is worth a thousand
words, how dangerous is a FAX? ...... About 85% of a GIF.

This time it will surely run.

I just found the last bug.

The generation of random numbers is too important to be left to chance.
-Robert R. Coveyou Oak Ridge National Laboratory

It's redundant! It's redundant! -R. E. Dundant

Bug? That's not a bug, that's a feature. -T. John Wendel

The programmer's national anthem is 'AAAAAAAAHHHHHHHH'. -Weinberg, p.152

Stack manipulation - the use of inflatable falsies.-Datamazing, 4/1/78

On a clear disk you can seek forever. -Computerworld button

I write all my critical routines in assembler, and my comedy routines in
FORTRAN. -Anonymous

If debugging is the process of removing bugs, then programming must be the
process of putting them in. -Dykstra

"#define QUESTION ((bb) || !(bb)) - Shakespeare."

"Real programmers use: COPY CON PROGRAM.EXE"

May the bugs of many programs nest on your hard drive.

I'm a modemer and I'm OK. I post all night and I sleep all day.

I modem, but they grew back.

Logic: The art of being wrong with confidence...

Logic is neither an art or a science but a dodge.

To iterate is human; to recurse, divine.

Do you like me for my brain or my baud?

If at first you don't succeed, call it version 1.0

Maintenance-free: When it breaks, it can't be fixed...

Memory dump: Amnesia...

Microwave: Signal from a friendly micro...

Modem: How a Southerner asks for seconds...

Nostalgia: The good old days multiplied by a bad memory...

WOMEN.ZIP: A great program, but it doesn't come with documentation...

WOMAN.ZIP: Great Shareware, but be careful of viruses...

29A, the hexadecimal of the Beast.

SET DEVICE=EXXON to screw up your environment.

My BBS is baroque now. Please call Bach later with your Handel.

This BBS is ancient. Some say from the echocene.

God is REAL, unless explicitly declared INTEGER.

Asking if computers can think is like asking if submarines can swim.

>From C:*.* to shining C:*.*

Nice computers don't go down.

Resistance is useless! (If AAAAAA - American Association Against Acronym
Abuse Anonymous

Me and my two friends... GIF and Wesson.

I'm not a sysop, I just play one on the echoes.

CCITT - Can't Conceive Intelligent Thoughts Today

This message transmitted on 100% recycled electrons.

Todays assembler command : EXOP Execute Operator

Justify my text? I'm sorry but it has no excuse.

Programming is an art form that fights back.

[If you can't hear me, it's because I'm in parentheses]

grep..grep..grep... (Frog with UNIX stuck in its' throat)

"Daddy, what does FORMATTING DRIVE C mean?"

All wiyht. Rho sritched mg kegtops awound?

My mail reader can beat up your mail reader.

My Go this amn keyboar oesn't have any 's.

My computer NEVER cras

Never forget: 2 + 2 = 5 for extremely large values of 2.

Nobody has ever, ever, EVER learned all of WordPerfect.

The world's coming to an end. Log off and leave in an orderly fashion.

To define recursion, we must first define recursion.

"The Soviet Union does not exist any more in its present format." CCCP:>
format CCCP: /u

---

The Really Hard Questions In Life

Why doesn't glue stick to the inside of the bottle?
Would a fly without wings be called a walk?
Why do they lock gas station bathrooms? Are they afraid someone will clean them?
Why do steam irons have a permanent press setting?
Can you be a closet claustrophobic?
Why do you need a driver's license to buy liquor, when you can't legally drink and drive?
Does the Postmaster General need a stamp of approval?
Why is the word abbreviation so long?
Is it possible to be totally partial?
What's another word for thesaurus?
If a book about failures doesn't sell, is it a success?
If the funeral procession is at night, do folks drive with their lights off?
When companies ship styrofoam, what do they pack it in?
If you're cross-eyed and have dyslexia can you read all right?
If a stealth bomber crashes in a forest, will it make a sound?
If the cops arrest a mime, do they tell him he has the right to remain silent?
If a parsley farmer is sued, can they garnish his wages?
If a turtle doesn't have a shell, is he homeless or naked?
When it rains, why don't sheep shrink?
Should vegetarians eat animal crackers?
Do cemetery workers prefer the graveyard shift?
What do you do when you see an endangered animal that eats only endangered plants?
Do hungry crows have ravenous appetites?
Why do people who know the least know it the loudest?
What if there were no rhetorical questions?

 

---

LETTERMAN'S TOP 10 THINGS THAT SOUND DIRTY AT THE OFFICE, BUT AREN'T

10. I need to whip it out by 5!
9. Mind if I use your laptop?
8. Put it in my box before you leave.
7. If I have to lick one more, I'll gag!
6. I want it on my desk, NOW!
5. Hmmmmm.....I think it's out of fluid.
4. My equipment is so old it takes forever to finish.
3. It's an entry-level position.
2. When do you think you'll be getting off today? AND NUMBER 1 IS...
1. Its not fair...I do all the work while he just sits there!

 

---

FOR WOMEN ONLY

TO BE USED FOR "MEDICAL" PURPOSES ON THOSE DAYS WHEN YOU ARE NOT EXACTLY OVERJOYED WITH YOUR PARTNER IN LIFE How many men does it take to change a roll of toilet paper?
We don't know, it's never happened.

How are men and parking spots alike?
All the good ones are taken and the ones that are left are handicapped.

What's a man's idea of helping with the housework?
Lifting his legs so you can vacuum.

What did God say after creating man?
"I can do better."

If men got pregnant, abortions would be available atconvenience stores and drive-through windows.
Why do men name their penises?
Because they want to be on a first name basis with the person who makes all the decisions.

Why is it so hard for women to find men who are sensitive, caring and good looking?
Because those men already have boyfriends.

Why do men like masturbation?
It's sex with someone they love.

Why were men given larger brains than dogs?
So they wouldn't hump women's legs at cocktail parties.

How does a man take a bubble bath?
He eats beans for dinner.

How can you tell if a man is sexually excited?
He's breathing.

What's the difference between men and government bonds?
Bonds mature.

What's the difference between a man and a catfish?
One is a bottom feeding scum sucker and the other one is a fish.

What do you have when you hold two balls in your hand?
A man's undivided attention.

Why are men like laxatives?
They irritate the shit out of you.

What's a man's view of safe sex?
A padded headboard.

How do men sort laundry?
"Filthy," and "Filthy but wearable."

What do you call a man who understands a little bit about women?
Gifted.

Church Entrance Exam

Three couples, an elderly couple, a middle aged couple and a young newlywed couple wanted to join a church. The pastor says, "We have special requirements for new parishioners. You must abstain from having sex for two weeks." The couples agreed and came back at the end of two weeks.

The pastor goes to the elderly couple and asks, "Were you able to abstain from sex for the two weeks?" The old manreplies, "No problem at all, Pastor." "Congratulations! Welcome tothe church." said the pastor. The pastor goes to the middle agedcouple and asks, "Well, were you able to abstain from sex for thetwo weeks?" The man replied, "The first week was not too bad. The secondweek I had to sleep on the couch for a couple of nights but, yes wemade it." "Congratulations! Welcome to the church." said the pastor. The pastorthen goes to the newlywed couple and asks, 'Well, were you able to abstainfrom sex for two weeks?"

"Well Pastor, we were not able to go without sex for the two weeks" the young man replied. "What Happened?" inquired the pastor. "My wife was reaching for a can of corn on the top shelf and dropped it. When she bent over to pick it up, I was over come with lustand took advantage of her right there." "You understand, of course, this means youwill not be welcome in our church." Stated the pastor. "That's OK."Said the young man, "We're not welcome at Safeway anymore either.

---

If I ever become an Evil Overlord

1. My legions of terror will have helmets with clear plexiglass visors, not face-concealing ones.
2. My ventilation ducts will be too small to crawl through.
3. My noble half-brother whose throne I usurped will be killed, not kept anonymously imprisoned in a forgotten cell of my dungeon.
4. Shooting is not too good for my enemies.
5. The artifact which is the source of my power will not be kept on the Mountain of Despair beyond the River of Fire guarded by the Dragons of Eternity. It will be in my safe-deposit box.
6. I will not gloat over my enemies' predicatment before killing them.
7. When the rebel leader challenges me to fight one-on-one and asks,"Or are you afraid without your armies to back you up?" My reply will be, "No, just sensible."
8. When I've captured my adversary and he says, "Look, before you kill me, will you at least tell me what this is all about?" I'll say, "No,"and shoot him.
9. After I kidnap the beautiful princess, we will be married immediately in a quiet civil ceremony, not a lavish spectacle in three weeks time during which the final phase of my plan will be carried out.
10. I will not include a self-destruct mechanism unless absolutely necessary. If it is necessary, it will not be a large red button labelled "Danger: Do Not Push".
11. I will not order my trusted lieutenant to kill the infant who is destined to overthrow me -- I'll do it myself.
12. I will not interrogate my enemies in the inner sanctum -- a small hotel well outside my borders will work just as well.
13. I will be secure in my superiority. Therefore, I will feel no need to prove it by leaving clues in the form of riddles or leaving my weaker enemies alive to show they pose no threat.
14. I will not waste time making my enemy's death look like an accident -- I'm not accountable to anyone and my other enemies wouldn't believe it.
15. I will make it clear that I _do_ know the meaning of the word "mercy"; I simply choose not show them any.
16. One of my advisors will be an average five-year-old child. Any flaws in my plan that he is able to spot will be corrected beforeimplementation.
17. All slain enemies will be cremated, not left for dead at the bottom of the cliff. The announcement of their deaths, as well asany accompanying celebration, will be deferred until after theaforementioned disposal.
18. My undercover agents will not have tattoos identifying them as members of my organization, nor will they be required to wear military boots or adhere to any other dress codes.
19. The hero is not entitled to a last kiss, a last cigarette, or any other form of last request.
20. I will never employ any device with a digital countdown. If I find that such a device is absolutely unavoidable, I will set it to active when the counter reaches 117 and the hero is just putting his plan into operation.
21. I will design all doomsday machines myself. If I must hire a mad scientist to assist me, I will make sure that he is sufficiently twisted to never regret his evil ways and seek to undo the damage he's caused.
22. I will never utter the sentence "But before I kill you, there's just one thing I want to know."
23. When I employ people as advisors, I will occasionally listen to their advice.

---

Smart House?

Tele-Communications Inc., the nation's largest cable television company, is in talks to launch a unique pilot project in conjunction with Pacific Gas & Electric Co. and Microsoft Corp. to design a "smart home." The home automation industry is expected to triple in size, from $1.7 billionthis year to more than $5.1 billion by the year 2000.

November 28, 1995 Moved in at last. Finally, we live in the smartest house in the neighborhood. Everything's networked. The cable TV is connected to our phone, which is connected to my personal computer, which is connected to the power lines, all the appliances and the security system. Everything runs off a universal remote with the friendliest interface I've ever used. Programming is a snap. I'm, like, totally wired.

November 30 Hot stuff! Programmed myVCR from the office, turned up the thermostat and switched onthe lights with the car phone, remotely tweaked the oven a few degreesfor my pizza. Everything nice and cozy when I arrived. Maybe I should get the universal remote surgically attached.

December 3 Yesterday, the kitchen crashed. Freak event. As I opened the refrigerator door, the light bulb blew. Immediately, everything else electrical shutdown- - lights, microwave, coffee maker - everything. Carefully unplugged and replugged all the appliances. Nothing. Called the cable company (but not from the kitchen phone). They refer me to the utility. The utility insists the problem was in the software. So the software company runs some remote telediagnostics via my house processor. Their expert system claims it has tobe the utility's fault. I don't care, I just want my kitchen back.More phone calls; more remote diagnostics. Turns out the problemwas "unanticipated failure mode" - the network had never seen a refrigerator bulb failure while the door was open. So the fuzzy logic interpreted the burnout as a power surge and shut down the entire kitchen. But because sensor memory confirmed that there hadn't actually been a power surge, the kitchen's logic sequence was confused so it couldn't do a standard restart. The utility guy swears this was the first time thishas ever happened. Rebooting the kitchen took over an hour.

December 7 The police are not happy. Our house keeps calling them for help. We discover that whenever we play the TV or stereo above 25 decibels, it creates patterns of micro-vibrations that get amplified when they hit the window. When these vibrations mix with a gust of wind, the security sensors are actuated, andthe police computer concludes that someone is trying to break in.Go figure. Another glitch: Whenever the basement is in self-diagnosticmode, the universal remote won't let me change the channels on myTV. That means I actually have to get up off the couch and change thechannels by hand. The software and the utility people say this flawwill be fixed in the next upgrade- SmartHouse 2.1. But it's not readyyet.

December 12 This is a nightmare. There's a virus in the house. My personal computer caught it while browsing on the public access network. I come home and the living room isa sauna, the bedroom windows are covered with ice, the refrigerator has defrosted, the washing machine has flooded the basement, the garagedoor is cycling up and down, and the TV is stuck on the home shopping channel. Throughout the house, lights flicker like stroboscopes until they explode from the strain. Broken glass is everywhere. Of course, the security sensors detect nothing. I look at a message slowly throbbingon my personal computer screen: "Welcome to HomeWrecker!!! Now the Fun Begins ... (Be it ever so humble, there's no virus like HomeWrecker ... )" I get out of the house. Fast.

December 18 They think they've digitally disinfected the house, but the place is a shambles. Pipes have burst and we're not completely sure we've got the part of the virus that attacks toilets. Nevertheless, the Exorcists (as the anti-virus SWAT members like to call themselves) are confident the worst is over. "HomeWrecker is pretty bad," one tells me, "but consider yourself lucky you didn't get PolterGeist. That one is really evil."

December 19 Apparently, our house isn't insured for viruses. "Fires and mudslides, yes," says the claims adjuster. "Viruses, no." My agreement with the SmartHouse people explicitly states that all claims and warranties are null and void if any appliance or computer in my house networks in any way, shape or form with a noncertified on-line service. Everybody'svery, very sorry, but they can't be expected to anticipate every virus that might be created. We call our lawyer. He laughs. He'sexcited.

December 21 I get a call from a SmartHouse sales rep. As a special holiday offer, we get the free opportunity to become a beta site for the company's new SmartHouse 2.1 upgrade. He says I'll be able to meet the programmers personally. "Sure," I tell him.

above written by: Aamir A. Rashid, The Open Group, Research Institute, Research Engineer, Eleven Cambridge Center, Cambridge, MA 02142

---

Personnel Department Placement Test

What is a human resource? Does your organization struggle with the problem of properly fitting people to jobs? Here is a handy hint for ensuring success in job placement. Take the prospective employees you are trying to place and put them in a room with only a table and two chairs. Leave them alone for two hours, without any instruction. At the end of that time, go back and see whatthey aredoing. If they have taken the table apart in that time, put themin Engineering. If they are counting the butts in the ashtray,assign them to Finance. If they are screaming and waving theirarms, send them off to Manufacturing. If they are talkingto the chairs, Personnel is a good spot for them. If theyare sleeping, they are Management material. If they are writingup the experience, send them to Tech Pubs. If they don't even look upwhen you enter the room, assign them to Security. If they tryto tell you it's not as bad as it looks, send them to Marketing. And if they've left early, put them in Sales.

---

Dumb Things People Wish They Hadn't Said

"I haven't committed a crime. What I did was fail to comply with the law"
David Dinkins, New York City Mayor, answering accusations that he failed to pay his taxes.

"They gave me a book of checks. They didn't ask for any deposits."
Congressman Joe Early (D-Mass) at a press conference to answer questions about the House Bank Scandal.

"He didn't say that. He was reading what was given to him in a speech."
Richard Darman, director of OMB, explaining why President Bush wasn't following up on his campaign pledge that there would be no loss of wetlands.

"It depends on your definition of asleep. They were not stretched out. They had their eyes closed. They were seated at their desks with theirheads in a nodding position."
John Hogan, Commonwealth Edison Supervisor of News Information, responding to a charge by a Nuclear Regulatory Commission inspector that two Dresden Nuclear Plant operators were sleeping on the job.

"I didn't accept it. I received it."
Richard Allen, National Security Advisor to President Reagan, explaining the $1000 in cash and two watches he was given by two Japanese journalists after he helped arrange a private interview for them with First Lady Nancy Reagan.

"I was a pilot flying an airplane and it just so happened that where I was flying made what I was doing spying."
Francis Gary Power, U-2 reconnaissance pilot held by the Soviets for spying, in an interview after he was returned to the US.

"I was under medication when I made the decision not to burn the tapes."
President Richard Nixon

"Smoking kills. If you're killed, you've lost a very important part of your life."
Brooke Shields, during an interview to become spokesperson for a federal anti-smoking campaign.

"I've never had major knee surgery on any other part of my body."
Winston Bennett, University of Kentucky basketball forward.

"I support efforts to limit the terms of members of Congress, especially members of the House and members of the Senate."
Vice-President Dan Quayle

"Outside of the killings, Washington has one of the lowest crime rates in the country."
Mayor Marion Barry, Washington, DC

"Sure, it's going to kill a lot of people, but they may be dying of something else anyway."
Othal Brand, member of a Texas pesticide review board, on chlordane.

"Are you any relation to your brother Marv?"
Leon Wood, New Jersey Nets guard, to Steve Albert, Nets TV commentator.

"Beginning in February 1976 your assistance benefits will be discontinued. Reason: it has been reported to our office that you expired on January1, 1976."
Letter from the Illinois Department of Public Aid

"The Holocaust was an obscene period in our nation's history... this century's history.... We all lived in this century. I didn't livein this century."
Dan Quayle, then Indiana senator and Republican vice-presidential candidate during a news conference in which he was asked his opinion of the Holocaust.

"In the early sixties, we were strong, we were virulent..."
John Connally, Secretary of Treasury under Richard Nixon, in an early seventies speech, as reported in a contemporary "American Scholar".

"Rotarians, be patriotic! Learn to shoot yourself."
Chicago Rotary Club journal, "Gyrator".

"The streets are safe in Philadelphia. It's only the people who make them unsafe." Frank Rizzo, ex-police chief and mayor of Philadelphia.

"I've always thought that underpopulated countries in Africa are vastly underpolluted."
Lawrence Summers, chief economist of the World Bank, explaining whywe should export toxic wastes to Third World countries.

"The crime bill passed by the Senate would reinstate the Federal death penalty for certain violent crimes: assassinating the President;hijacking an airliner; and murdering a government poultry inspector."
Knight Rider News Service dispatch

"After finding no qualified candidates for the position of principal, the school board is extremely pleased to announce the appointmentof David Steele to the post."
Philip Streifer, Superintendent of Schools, Barrington Rhode Island.

"The doctors X-rayed my head and found nothing."
Dizzy Dean, explaining how he felt after being hit on the head by aball in the 1934 World Series.

---

Ten Myths Perpetrated by American Movies

#1. The entire population of France speaks fluent, charming English.
#2. A six-shooter holds 18 bullets.
#3. Breathtaking women regularly work as juke joint waitresses, dog walkers, supermarket cashiers, pump jockeys, and police detectives. #4. Any injury suffered in a bar fight (including extracted teeth) takes no more than six minutes to heal.
#5. All priests are musical.
#6. Convenient parking spaces are readily available in major American cities any time of day or night.
#7. Men can spend weeks on a raft or in a desert and never grow more than a two day stubble of beard.
#8. Anyone can down six or seven shots of straight whiskey and then get a gun out of a holster.
#9. Dogs, horses, whales and pigs sport senses of humor that rival Noel Coward's.
#10. Money doesn't matter. written by Joe Bailey, New York City

---

HARVARD INTERDISCIPLINARY STUDIES FINAL EXAM

COMPUTER SCIENCE: Write a fifth-generation computer language. Using that language, write a program to dothe rest of this examination.
HISTORY: Describe the history of the papacy from its origins to the present, concentrating on its social, political, economic, religious, and philosophical impact on Europe, America, Asia,and Africa. Be concise and specific.
ELECTRICAL ENGINEERING: You will be placed in a nuclear reactor and given a wiring diagram. The electrical system has been tampered with. You have seventeen minutes to find the problem and correctit before a meltdown begins.
MEDICINE: You will be provided with a Swiss Army knife, some gauze, a needle and thread, and a bottle of Scotch. Remove your appendix. Do not suture until your work has been inspected. You have 15 minutes.
PUBLIC SPEAKING: 2500 torch-bearing Pro-Life crazies are storming the clinic. Calm them while performing an abortion.
BIOLOGY: Create life. Estimate the differences in our culture if this life form had developed 500,000 years earlier, with special attention to the probable effect, if any, on the Snail Darter.
CIVIL ENGINEERING: By inter-weaving toothpicks, build a platform that will support your weight when suspended over a vat of concentrated sulfuric acid. Field-test the platform under the conditions for which it was designed.
MUSIC: Write a piano concerto blending the styles of early J. S. Bach and late Igor Stravinsky that could win a competition judged by Josef Stalin and John Cage. Orchestrate it as Ravel might have. Perform the solo part. You will find a piano under your seat.
PSYCHOLOGY: Based on your knowledge of their writings, evaluate the emotional stability, degree of adjustment, and repressed frustrations of Alexander of Aphrodisias, Ramses II, and Gregoryof Nicea. Support your evaluationwith untranslated quotations fromtheir work.
CHEMISTRY: A poison is in one beaker on your laboratory table; its antidote is in an identical one. Analyze them, determine which is which, and drink them. Note: if the wrong beaker is drunk first, you will die.
SOCIOLOGY: Estimate the sociological problems associated with the end of the world. Construct an experiment to test your answer using double-blind methodology.
MECHANICAL ENGINEERING: The disassembled parts of an automatic pistol are in a box on your desk along with an instruction manual printed in Machine Language. In ten minutes a starved tiger will be admitted to the room. Take whatever action you feel appropriate.Cite provisions of the National Building Code to justify your action.
ECONOMICS: Describe and contrast the effects of interlocking directorates (be specific) and child-labor laws on the causes of the Great Depression of the 1930s.
MATHEMATICS: Derive the Euler-Cauchy equations using a straightedge, a compass, and a pencil. Discuss in detail the role these equations had on mathematical analysis in 1920, 1960, and this year.
POLITICAL SCIENCE: There is a red telephone on your desk. Start and end World War III. Describe its socio-political effects onBurma.
RELIGION: Using only rational discourse and three diagrams, convert the Muslim Palestinian provided you to Judaism.
ART: Given crayons and a ream of paper, recreate the ceiling of the Sistine Chapel with particular regard for color accuracy.
PHYSICS: Choose one: Explain the nature of matter; evaluate in depth the impact of mathematics on science; derive the equations for anti-gravity; invent and build a magnet that attracts wood.
METAPHYSICS: Describe in detail the probable nature of life before conception and after death. Test your hypothesis.
PHILOSOPHY: Trace the development of the major and minor western and eastern moral theories and discuss the impact on free willeach has had. Discuss it a second time from a contrary viewpoint.Discuss it a third time from a contradictory viewpoint. Using Aristotelianlogic, prove each view point to be unassailable.
GENERAL KNOWLEDGE: Write a critique on the sum total of general knowledge point by point.
EXTRA CREDIT: Define a universe, and contrast two in detail.

---

Physics Story

A true story.
A thermodynamics professor had written a take home exam for his graduate students. It had one question: Is hell exothermic or endothermic? Support your answer with a proof. Most of the students wrote proofs of their beliefs using Boyle's Lawor some variant. One student, however wrote the following:

First, we postulate that if souls exist, then they must have some mass. If they do, then a mole of souls can also have a mass. So, at what rate are souls moving into hell and at what rate are souls leaving? I think that we can safely assume that once a soul gets to hell, it will not leave. Therefore, no souls are leaving. As for souls entering hell, lets look at the different religions that exist in the world today. Some of these religions state that if you are not a member of their religion, you will go to hell. Since there are more than one of these religions and peopledo not belong to more than one religion, we can project that all peopleand all souls go to hell. With birth and death rates as they are, we can expect the number of souls in hell to increase exponentially.Now, we look at the rate of change in volume in hell. Boyle's Law statesthat in order for the temperature and pressure in hell to stay the same, the ratio of the mass of souls and volume needs to stay constant.So, if hell is expanding at a slower rate than the rate at which soulsenter hell, then the temperature and pressure in hell will increaseuntil all hell breaks loose. Of course, if hell is expanding at a rate faster than the increase of souls in hell, than the temperature andpressure will drop until hell freezes over.

It was not revealed what grade the student got.

---

Politics and cows

FEUDALISM
You have two cows. Your lord takes some of the milk.
PURE SOCIALISM
You have two cows. The government takes them and puts them in a barn with everyone else's cows. You have to take care of all the cows. The government gives you a glass of milk.
BUREAUCRATIC SOCIALISM
Your cows are cared for by ex-chicken farmers. You have to take care of the chickens the government took from the chicken farmers. The government gives you as much milk and eggs the regulations say you should need.
FASCISM
You have two cows. The government takes both, hires you to take care of them, and sells you the milk.
PURE COMMUNISM
You share two cows with your neighbors. You and your neighborsbicker about who has the most "ability" and who has the most "need". Meanwhile, no one works, no one gets any milk, and the cows drop dead of starvation.
RUSSIAN COMMUNISM
You have two cows. You have to take care of them, but the government takes all the milk. You steal back as much milk as you can and sell it on the black market.
PERESTROIKA
You have two cows. You have to take care of them, but the Mafia takes all the milk. You steal back as much milk as you can and sell it on the "free" market.
CAMBODIAN COMMUNISM
You have two cows. The government takes both and shoots you.
DICTATORSHIP
You have two cows. The government takes both and drafts you.
PURE DEMOCRACY
You have two cows. Your neighbors decide who gets the milk.
REPRESENTATIVE DEMOCRACY
You have two cows.Your neighbors pick someone to tell you who gets the milk.
BUREAUCRACY
You have two cows. At first the government regulates what you can feed them and when you can milk them. Then it pays you not to milk them. Then it takes both, shoots one, milks the other and pours the milk down the drain. Then it requires you to fill out forms
accounting for the missing cows.
CAPITALISM
You don't have any cows. The bank will not lend you money to buy cows, because you don't have any cows to put up as collateral.
PURE ANARCHY
You have two cows. Either you sell the milk at a fair price oryour neighbors try to take the cows and kill you.
ANARCHO-CAPITALISM
You have two cows. You sell one and buy a bull.
SURREALISM
You have two giraffes. The government requires you to take harmonica lessons.
OLYMPICS-ISM
You have two cows, one American, one Chinese. With the help oftrilling violins and state of the art montage photography, John Tesh narrates the moving tale of how the American cow overcame the agony of growing up in a suburb with (gasp) divorced parents, then mentions in passing that the Chinese cow was beaten every dayby a tyrannical farmer and watched its parents butchered before itseyes. The American cow wins the competition, severely sprainingan udder in a gritty performance, and gets a multi-million dollar contractto endorse Wheaties. The chinese cow is led out of the arena and shot byChinese government officials, though no one ever hears about it.McDonald's buys the meat and serves it hot and fast at its Beijing restaurant.

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One Liners

1. The trouble with being punctual is that nobody's there to appreciate it.
-- Franklin P. Jones

2. Women's creed: Men are like linoleum. If you lay them right the first
time, you can walk on them for 20 years.

3. Love is the answer, but while you are waiting for the answer sex raises some pretty good questions.
-- Woody Allen

4. Lord, grant me the serenity to accept the things I can not change, the courage to change the things I can, and the wisdom to hide the bodies of Those I had to kill because they pissed me off.

5. I think animal testing is a terrible idea; they get all nervous and give the wrong answers.

6. All my life, I always wanted to be somebody. Now I see that I should Have been more specific.
-- Jane Wagner

7. The hypothalamus is one of the most important parts of the brain, Involved in many kinds of motivation, among other functions. The hypothalamus Controls the "Four F's": 1. fighting; 2. fleeing; 3. feeding; and 4.mating.
-- Psychology professor in neurophysiology intro course

8. Bigamy is having one wife too many. Monogamy is the same.
-- Oscar Wilde

9. What is a committee? A group of the unwilling, picked from the unfit,
To do the unnecessary.
-- Richard Harkness, The New York Times, 1960

10. I am not a vegetarian because I love animals; I am a vegetarian because I hate plants.
-- A. Whitney Brown

11. Experience is that marvelous thing that enables you recognize a mistake when you make it again.
-- F. P. Jones

12. Human beings, who are almost unique in having the ability to learn from the experience of others, are also remarkable for their apparent disinclination to do so.
-- Douglas Adams

13. As your attorney, it is my duty to inform you that it is not important that you understand what I'm doing or whyyou're paying me so much money. What's important is that you continueto do so.
-- Hunter S. Thompson's Samoan Attorney

14. Life may have no meaning. Or even worse, itmay have a meaning of which I disapprove.
-- Ashleigh Brilliant

15. My opinions may have changed, but not the fact that I am right.
-- Ashleigh Brilliant

---

Stranded Engineers

There was this male engineer, on a cruise ship in the Caribbeanfor the first time. It was wonderful, the experience of his life. He was being waited on hand an foot. But, it did not last. A hurricane cameup unexpectedly. The ship went down almost instantly.

The man found himself, he knew not how, swept up on the shore of an island. There was nothing else anywhere to be seen. No person, no supplies, nothing. The man looked around. There were some bananas and coconuts, but that was it. He was desperate, and forlorn, but decided to make the bestof it. So for the next four months he ate bananas, drank coconut juice andmostly looked to the seamightily for a ship to come to his rescue.

One day, as he was lying on the beach stroking his beard and looking for a ship, he spotted movement out ofthe corner of his eye. Could it be true, was it a ship? No, fromaround the corner of the island came thisrowboat. In it was themost gorgeous woman he had ever seen, or at leastseen in 4 months.She was tall, tanned, and her blond hair flowing in thesea breeze gaveher an almost ethereal quality. She spotted him also ashewasaving and yelling and screaming to get her attention. She rowed her boattowardshim.

In disbelief, he asked, "Where did you come from? How did you get here"? She said, "I rowed from the other side of the island. I landed onthis island when my cruise ship sank" "Amazing," he said, "I didn't knowanyone else had survived. Howmany of you are there? Where, did you get therowboat? You musthave been really lucky to have a rowboat wash up with you?"

"It is only me,"she said, "and the rowboat didn't wash up, nothing else did." "Well then," said the man,"how did you get the rowboat?" "I made the rowboat out of raw materialthat I found on the island," replied the woman. "The oarswere whittled from Gum tree branches, I wove the bottom from Palmbranches, and the sides and stern came from a Eucalyptus tree".

"But, but," asked the man, "what about tools and hardware, how did you do that?" "Oh, no problem," replied the woman, "on the south side of the island there is a very unusual strata of alluvial rock exposed. Ifound that if I fired it to a certain temperature in mykiln, it meltedinto forgeable ductile iron. I sed that for tools, and used the toolsto make the hardware. But, enough of that," she said."Where do you live?"

At last the man was forced to confess that he had been sleepingon the beach. "Well, let's row overto my place," she said. So theyboth got into the rowboatand left for her side of island. The woman easily rowed them aroundto a wharf that led to the approach to her place.She tied up the rowboatwith a beautifully woven hemp rope. They walked up a stone walk
and around a palm tree, there stood an exquisite bungalow painted in blue and white.

"It's not much," she said, "but I call it home. Sit down please, would you like to have a drink?" "No," saidthe man, "one more coconut juice and I will puke." "It won't be coconut juice," the woman replied, "I have a still, how about a Pina Colada?" Trying to hide his continued amazement, the man accepted, and they sat down on her couch to talk.

After a while, and they had exchanged their stories, the woman asked, "Tell me, have you always had a beard?" "No," the man replied, "I was clean shaven all of my life, and even on the cruise ship." "Well if you wouldlike to shave, there is a man's razor upstairs in the cabinet in the bathroom." So, theman, no longer questioning anything, went upstairs to the bath room.There in the cabinet was a razor made from a bone handle, two shellshoned to a hollow ground edge were fastened on to its end inside of a swivel mechanism. The man shaved, showered and went back down stairs.

"You look great," said the woman, "I think I will go up and slip into something more comfortable." So shedid. And, the man continued to sip his Pina Colada. After ashort time, the woman returned wearing fig leafs strategically positionedand smelling faintly of gardenia. "Tell me," she asked, "we have bothbeen out here for a very long time with no companionship. You know whatI mean. Have you been lonely, is there anything that you really miss?Something that all men and women need. Something that it would be really nice to have right now."

"Yes there is," the man replied, as he moved closer to the woman while fixing a winsome gaze upon her. "Tell me ... Do you happen to have anInternet connection?"

---

IDEAS FOR THE WORKPLACE

Put a chair facing a printer, sit there all day and tell people you're waiting for your document. Insist that your e-mailaddress is zena_goddess_of_fire@companyname.com"

Every time someone asks you to do something, ask them tosign a waiver

Every time someone asks you to do something, ask them if they want fries with that.

Page yourself over the intercom. (Don't disguise your voice.)

Name all your pens and insist that meetings can't begin until they're all present.

Find out where your boss shops and buy exactly the same outfits. Always wear them one day after your boss does. (This is especially effective if your boss is a different gender than you are.)

Make up nicknames for all your coworkers and referto them only by these names: "That's a good point Sparky."; "No I'm sorry I'm going to have to disagree with you there, Chachi."

Include a piece of your children's artwork as a cover page for all reports that you write. (If you don't have children, draw stick figures yourself.)

Agree to organize the company Christmas party. Hold it at McDonald's Playland. Charge everyone $15 each.

Send email to the rest of the company telling them what you're doing. For example "If anyone needs me I'll be in the bathroom."

No matter what anyone asks you, reply "Okay."

Put your garbage can on your desk. Label it "IN."

Develop an unnatural fear of staplers.

Install a set of buttons and lights in the arm of your chair.

Talk into your daytimer.

Email nude gifs (graphic image files) of yourself to your coworkers. Tell them you got them off the Internet.

While sitting at your desk, soak your fingers in "Palmolive."

Put up mosquito netting around your cubicle.

Decorate your office with pictures of Cindy Brady and Danny Partridge.Try to pass them off as your children. Send e-mail messages saying free pizza, free donuts etc... in the lunchroom, when people complain that there was none... Just lean back, pat your stomach, and say, "Oh you've got to be faster than that."

---

THE COSTUME PARTY

A couple was invited to a swanky masked Halloween Party. The wife came down with a terrible headache and told her husband to go to the party alone. He, being a devoted husband, protested, butshe argued and said she was going to take some aspirin and go to bed,and there was no need of his good time being spoiled by not going. Sohe took his costume and away he went.

The wife, after sleeping soundly for one hour, awakened without pain, and as it was still early, she decided to go to the party. In as much as her husband did not know what her costume was, she thought she would have some fun by watching her husband to see how he acted when she was not with him.

She joined the party and soon spotted her husband cavorting around on the dance floor, dancing with every woman he could find, and copping a little feel here and a little kiss there. His wife sidled up to him and being a rather seductive babe herself, he left his partner high and dry and devoted his time to the new stuff that had just arrived.

She let him go as far as he wished; naturally, since he was her husband. Finally he whispered a little proposition in her ear and she agreed, so off they went to one of the cars and had a little fun. Just before unmasking at midnight, she slipped away and went home and put the costume away and got into bed, wondering what kind of explanation he would make for his behavior.

She was sitting up reading when he came in and asked what kind of a time he had. He said, "Oh, the same old thing. You know Inever have a good time when you're not there." Then she asked, "Did you dancemuch?"

He replied, "I'll tell you, I never even dancedone dance. When I got there, I met Pete, Bill Brown and some other guys,so we went into the den and played poker all evening. But I'll tell you...the guy I loaned my costume to sure had a real good time!"

Read the manual...

Well, after the big bad wolf had been killed by the woodsman, his cousin, big bad nasty wolf moved into the forest. Grandmother had heard about him and she warned Little Red Riding Hood about staying on the pathwhile walking to her house. One day, Mother prepared freshly baked breadfor Little Red Riding Hood to deliver to Grandmother. On her way, as sheskipped down the path, the big bad nasty wolf jumped out onto the pathand said, "Give me the basket of goodies, or I'm going to pull down yourlittle red panties and f--- your little red socks off." Immediately, LittleRed Riding Hood handed over the basket and took off running. Mother wasvery upset with Little Red Riding Hood for not delivering the basket toGrandmother, even though Little Red Riding Hood didn't tell Mother whathad happened; she thought she could take care of this herself.She put herfather's .38 special inside the basket under the new loaf of
bread.

So, the next day, while skipping along the pathto Grandmother's house, the big bad nasty wolf jumped out in front of her and said, "Hand the basket over, or I'm going to pull down your little red panties and f--- your little red socks off." This time though, Little Red Riding Hood pulled out the pistol, pointed it straight at the wolf and replied, "But first, you're going to eat me, like the book says."

 

---

From a book called, "Wisdom From The Walls," by Kristen Kammerer and
Bridget Snyder.

They compiled some really outstanding graffiti, and here it is:

The best way to a man's heart is to saw his breast plate open. * Women's restroom, Murphy's, Champaign, IL

Don't trust anything that bleeds for 5 days and doesn't die. * Men's restroom, Murphy's, Champaign, IL

If you can piss this high, join the fire department. On the wall in the men's restroom at a height of 6 feet. * O'Ryan's -Irish Pub. Ashland, Oregon.

Beauty is only a light switch away. * Perkins Library. Duke University. Durham, North Carolina.

I've decided that to raise my grades I must lower my standards. * Houghton Library, Harvard University. Cambridge, Massachusetts.

Remember, it's not, "How high are you?" it's "Hi, how are you?" * Rest stop off Route 81. West Virginia.

God made pot. Man made beer. Who do you trust? -The Irish Times. Washington, D.C.

Fighting for peace is like screwing for virginity. * The Bayou, Baton Rouge, Louisiana.

No matter how good she looks, some other guy is sick and tired of putting up with her shit. * Men's Room, Linda's Bar and Grill. Chapel Hill, North Carolina.

It's hard to make a comeback when you haven't been anywhere. * Written in the dust on the back of a bus. Wickenburg, Arizona.

Make love, not war.-Hell, do both, get married! * Women's restroom, The Filling Station. Bozeman, Montana.

If voting could really change things, it would be illegal. * Revolution Books. New York, New York.

A Woman's Rule of Thumb: If it has tires or testicles, you're going to have trouble with it. * Women's restroom, Dick's Last Resort. Dallas, Texas.

JESUS SAVES! But wouldn't it be better if he had invested? *Men's restroom, American University. Washington, D.C.

If pro is opposite of con, then what is the opposite of progress? Congress! * Men's restroom, House of Representatives. Washington, D.C.


Express Lane: Five beers or less * Sign over one of the urinals, Ed Debevic's. Pheonix, AZ.

 

---

The Trucker

A trucker who had driven his fully loaded rig to the top of a steep hill and was just starting down the equally steep other side when he noticed a man and a woman lying in the center road,making love. He blew his airhorn several times as he was bearing down on them. Realizing that they werenot about to get out of his way he slammed on his brakes and stopped justinches from them.

Getting out of the cab, mader than hell, the trucker walked to the front of the cab and looked down at the two, still in theroad, and yelled, "What the hell's the matter with you two? Didn't you hearme blowing the horn? You could've been killed!"

The man on the highway, obviously satisfied and not too concerned, looked up and said, "Look, I was coming, she was coming, and you were coming. You were the only one with brakes."

 

---

BUMPER STICKERS

Auntie Em. Hate you, hate Kansas, taking the dog - Dorothy We're staying together for the sake of the cats.

It's been lovely, but I have to scream now.

My karma ran over your dogma.

Women who seek to be equal to men lack ambition.

This is not an abandoned vehicle.

I don't lie, cheat or steal unnecessarily.

Beautify Texas - Put a Yankee on a bus.

Welcome to Texas, now go home.

It's as bad as you think and they are out to get you.

If you don't like the news, go out and make some of your own.

My wife says if I go fishing one more time, she's going to leave me. Gosh, I'm going to miss her.

I is a college student.

Beer isn't just for breakfast any more.

Sorry, I don't date outside my species.

Eschew obfuscation.

Will Rogers never met a lawyer.

Happiness is seeing your mother-in-law's face on the back of a milk carton.

It's lonely at the top, but you eat better.

Don't steal - The government hates competition.

Is there life before coffee?

Never play leap frog with a unicorn.

Cover me - I'm changing lanes.

The weather is here - Wish you were beautiful.

I Cayman went.

My other wife is beautiful.

I need someone really bad - Are you really bad?

Smile - It's the second best thing you can do with your lips.

Don't laugh - Your daughter could be in this vehicle.

Geez if you belive in honkus.

Friends don't let friends drive naked.

Save California; when you leave take someone with you.

I came, I saw, I did a little shopping.

There's one in every crowd and they always find me.

If money could talk, it would say goodbye.

When you're in love, you're at the mercy of a stranger.

Just when you think you've won the rat race along come faster rats.

If it's too loud, you're too old.

Wink - I'll do the rest.

The worst day fishing is better than the best day working.

An Irishman is not drunk so long as he can hold on to one blade ofgrass and not fall off the earth.

Cynics are people who know the price of everything and the value of nothing.

Who cares who's on board?

No radio - Already stolen.

Crime wouldn't pay if the government ran it.

Want a taste of religion? Bite a minister.

Carls bad Caverns: 22% more cavities.

Honk if you love cheeses.

Flying saucers are real, the Air Force doesn't exist. I don't carewho you are, what you are driving, or where you would rather be.

So many pedestrians, so little time.

Honk if you're illiterate.

My kid can beat up your honor student

Fight crime, shoot back

Guns don't kill people postal workers do.

It's not how you pick your nose, it's where you put the boogers.

It's not how you pick it, but where you flick it

If you don't like the way I drive, get off the sidewalk!

Grow your own dope, plant a man.

My wife's other car is a broom

Go ahead and honk - I'm reloading

This car is like my husband, if it ain't yours don't touch it!

Give Blood Play Hockey

I like cats, they taste just like chicken

Fleece on earth, good wool to ewe

 

---

Top Twenty Engineering Definitions

1. A NUMBER OF DIFFERENT APPROACHES ARE BEING TRIED
We're still guessing.....
2. AN EXTENSIVE REPORT IS BEING PREPARED FOR A FRESH APPROACH TO THE PROBLEM
We just hired three kids right out of college.
3. CLOSE PROJECT COORDINATION
We know who to blame.
4. A MAJOR TECHNOLOGICAL BREAKTHROUGH
It works okay, but looks very high tech.
5. CUSTOMER SATISFACTION IS ASSURED ON DELIVERY
We are so far behind schedule the customer will be happy to get it
delivered.
6. PRELIMINARY OPERATIONAL TESTS WERE INCONCLUSIVE
The darn thing blew up when we turned it on.
7. TEST RESULTS WERE EXTREMELY GRATIFYING
We're surprised the stupid thing works.
8. THE ENTIRE CONCEPT WILL HAVE TO BE ABANDONED
The only person who understood the thing quit.
9. IT'S STILL IN PROCESS
It's so wrapped up in red tape that the situation is hopeless.
10. WE'LL LOOK INTO IT
Forget it! We have enough problems for now.
11. PLEASE READ AND INITIAL
Let's spread responsibility around for the screw up.
12. GIVE US THE BENEFIT OF YOUR THINKING
We'll listen to what you have to say as long as it doesn't interfere
with what we've already done.
13. GIVE US YOUR INTERPRETATION
I can't wait to hear this baloney...
14. SEE ME, LET'S DISCUSS IT
Come into my office, I'm lonely.
15. ALL NEW
Parts not interchangeable with the previous design.
16. RUGGED
Too heavy to lift!
17. LIGHTWEIGHT
Lighter than RUGGED.
18. YEARS OF DEVELOPMENT
One finally worked.
19. ENERGY SAVING
Savings are achieved when the power switch is off.
20. LOW MAINTENANCE
Impossible to fix if broken.

 

---

LUCKY

A guy is strolling down the street in Chicago where he comes across an old lamp. He picks it up, rubs it vigorously, and out pops a genie. Thegenie offers to grant him one wish, to which the guy replies "I've always wanted to be lucky." The genie grants his wish. So off the guy strolls,wondering how this will change his life, when he spies 10 dollars on thesidewalk. Not a bad start he thinks. As he picks it up, he notices a bettingshop across the road. He strolls over, looks through the racing lists, andsees a horse named Lucky Lad at 100/1 in the 4th at the Meadowlands. Heputs the 10 dollars on the horse to win, and what do you know, the horsebolts in. Feeling on a bit of a roll, he heads to the local illegal casino,fronts up at the roulette table and puts the whole 1010 dollars on "Luckyseven." Round and round the wheel spins, and "bang!" - Lucky Seven.Now he's really flying....what better way to celebrate than to head to thelocal brothel for a bit of horizontal folk dancing.

He knocks and enters, and as he does so he is showered with streamers and handed a glass of champagne. The madam of the establishment puts her arm around him and says, "Welcome sir! We have much pleasure in informing you that you are our lucky 1000th customer, and you have won the right to enjoy the pleasures on offer from any girl who works here, absolutely free of charge." The guy says that he's always fancied making it with an Indian girl....so he's ushered into one of the rooms when in strolls the mostgorgeous sub-continental he has ever seen. Not much time passes before clothingis strewn around the room and the Karma Sutra (pp 101 to 532) is being welland truly tested. At one point the guy pauses and says to the girl, "Youare one of the most beautiful women I've ever seen in my life. I can't believehow lucky I am. But there is one thing I don't really like about Indian women.I don't like that red spot that you all have on your forehead." The Indiangirl looks him in the eye and says, "Sir, I am here to please you and succumbto your every desire. If you wish to see it gone, then please scratch offmy caste mark." So the guy goes at it with his fingernail. All of a suddenhe leans back and starts killing himself laughing. "What's wrong, what'swrong?" asks the Indian girl. The guy replies, "You're never going to believethis, but I've just won a car!"

 

---

The Gynocologist

A Gynocologist decides to that he wants to change occupations and be an auto mechanic. So he goes to auto mechanic school. It comes time forthe final exam. He takes the exam and when he gets the test back he's shocked at the test score he received *200%*, so he goes to talk to the instructor. The instructor tells him he gave him 50% for taking the engine apart correctly, 50% for putting it back together correctly, and an extra 100% for doingit all through the muffler.

 

---

FOOD AND HEAVEN
- Submitted by Marilyn Bond
---------------------------------
This 85 year old couple, having been married almost 60 years, had died in a car crash. They had been in good health the last tenyears mainly due to her interest in health food, and exercise. When theyreached the pearly gates, St. Peter took them to their mansion which wasdecked out with a beautiful kitchen and master bath suite and Jacuzzi. Asthey "oohed and aahed" the old man asked Peter how much all this was goingto cost. "It's free," Peter replied, "this is Heaven."

Next they went out back to survey the championship golf course that the home backed up to. They would have golfing privileges everyday and each week the course changed to a new one representing the great golf courses on earth. The old man asked, "what are the green fees?". Peter's reply, "This is heaven, you play for free." Next they went to the club house and saw the lavish buffet lunch with the cuisines of the world laid out. "Howmuch to eat?" asked the old man. "Don't you understand yet? This isheaven, it is free!" Peter replied with some exasperation. "Well, where arethe low fat and low cholesterol tables?" the old man asked timidly. Peter lectured, "That's the best part...you can eat as much as you like of whatever you like and you never get fat and you never get sick. This is Heaven."

With that the old man went into a fit of anger, throwingdown his hat and stomping on it, and shrieking wildly. Peter and his wifeboth tried to calm him down, asking him what was wrong. The old manlooked at his wife and said, "This is all your fault. If it weren'tfor your blasted bran muffins, I could have been here ten years ago!"

 

---

WHAT PERSONAL ADS REALLY MEAN
- Submitted by Orestes Papazisis
--------------------------------------
OK everybody, it's the weekend and many of you may be going out with people you have met from personal ads - whether from the newspapers or from the Web. Well here are guidelines of what those code words in your date's personal ads really mean. While you're at it, print this out and bring it on the date. It could be fun :)


A LIST OF ABBREVIATIONS
"WOMEN SEEKING MEN" Classifieds
------------------------------
CODE WORD..... MEANS

40-ish 48
Adventurer Has had more partners than you ever will
Affectionate Possessive
Artist Unreliable
Average looking You figure this one out
Beautiful Pathological liar
Commitment-minded Pick out curtains, now!
Communication important Just try to get a word in edgewise
Contagious Smile Bring your penicillin
Educated College dropout
Emotionally Secure Medicated
Employed Has part-time job stuffing envelopes at home
Enjoys art and opera Snob
Enjoys Nature Bring your own granola
Exotic Beauty Would frighten a Martian
Feminist A Ball buster
Financially Secure One paycheck from the street
Free spirit Substance abuser
Friendship first Trying to live down reputation as slut
Fun Annoying
Gentle Comatose
Good Listener Hard to pull a word from her
Humorous Caustic
Intuitive Your opinion doesn't count
In Transition Needs new sugar-daddy to pay the bills
Light drinker Lush
Looks younger If viewed from far away in bad light
Loves Travel If you're paying
Loves Animals Cat lady
Mature Will not let you treat her like a farm
animal in bed, like last boyfriend did
New-Age All body hair, all the time
Non-traditional Ex-husband lives in the basement
Old-fashioned Desperate
Outgoing Loud
Passionate Loud
Poet Depressive Schizophrenic
Professional Bitch
Redhead Shops on the Clairol aisle
Reliable Frumpy
Reubenesque You can figure this one out
Romantic Looks better by candle light
Self-employed Jobless
Smart Insipid
Special Rode the small schoolbus w/ tinted windows
Spiritual Involved with a cult
Stable Boring
Tall, thin Anorexic
Tan Wrinkled
Weight proportional to You canfigure this one out as well
height
Wants Soulmate One step away from stalking
Widow Nagged first husband to death
Writer Pompous
Young at heart How about the rest
****************************************************************

A LIST OF ABBREVIATIONS
"MEN SEEKING WOMEN" Classifieds

CODE WORD... MEANS...

40-ish 52 and looking for 25-yr-old
Affectionate Needy andlooking for mother-figure
Artist Delicateego badly in need of massage
Athletic Sits on the couch andwatches ESPN
Average looking Unusual hair growth on ears, nose, and back
Distinguished-looking Too much grey
Educated Will always treat you like an idiot
Employed On management track at Radio Shack
Financially Secure I will spend some money on you,in return
for whichI will expect you to obey my
every whim for the duration of your
mortal life.
Free Spirit Sleeps with your sister
Friendship first As long asfriendship involves nudity
Fun Good with a remote and a six pack
Good looking Arrogant bastard
Honest Pathological Liar
Huggable Overweight, more body hair than Gentle Ben
ISO Slim, attractive female Would be better off with a labrador
retriever
Light drinker Headed for AA
Like to cuddle Insecure, overly dependent
Like romantic walks on I read Cosmo and think this is what you
the beach want to hear
Mature Until you get to know him
Open-minded Wants to sleep with your sister but she's
not interested
Physically fit I spend a lot of timein front of mirrors
admiring myself
Poet Once wrote on a bathroom stall
Professional Owns a white button down
Reliable Shows up on time--give or take 3 hours
Self-employed Same as for women and eat nachos all
weekend
Sensitive Needy
Smart Thinks Cheers is "the wittiest show ever
on TV"
Spiritual Once went to church with his grandmother
on Easter
Stable Occasional stalker, but never arrested
Thoughtful Says "Please" when demanding a beer
Virile Can read3 Penthouse Forums without
passing out
Young at heart Pedophile

 

---

We take you now to the Oval Office.

George: Condi! Nice to see you. What's happening?
Condi: Sir, I have the report here about the new leader of China.
George: Great. Lay it on me.
Condi: Hu is the new leader of China.
George: That's what I want to know.
Condi: That's what I'm telling you.
George: That's what I'm asking you. Who is the new leader of China?
Condi: Yes.
George: I mean the fellow's name.
Condi: Hu.
George: The guy in China.
Condi: Hu.
George: The new leader of China.
Condi: Hu.
George: The Chinaman!
Condi: Hu is leading China.
George: Now whaddya' asking me for?
Condi: I'm telling you Hu is leading China.
George: Well, I'm asking you. Who is leading China?
Condi: That's the man's name.
George: That's who's name?
Condi: Yes.
George: Will you or will you not tell me the name of the new leader of
China?
Condi: Yes, sir.
George: Yassir? Yassir Arafat is in China? I thought he was in the
Middle East.
Condi: That's correct.
George: Then who is in China?
Condi: Yes, sir.
George: Yassir is in China?
Condi: No, sir.
George: Then who is?
Condi: Yes, sir.
George: Yassir?
Condi: No, sir.
George: Look, Condi. I need to know the name of the new leader of China.
Get me the Secretary General of the U.N. on the phone.
Condi: Kofi?
George: No, thanks.
Condi: You want Kofi?
George: No.
Condi: You don't want Kofi.
George: No. But now that you mention it, I could use a glass of milk.
And then get me the U.N.
Condi: Yes, sir.
George: Not Yassir! The guy at the U.N.
Condi: Kofi?
George: Milk! Will you please make the call?
Condi: And call who?
George: Who is the guy at the U.N?
Condi: Hu is the guy in China.
George: Will you stay out of China?!
Condi: Yes, sir.
George: And stay out of the Middle East! Just get me the guy at the U.N.
Condi: Kofi.
George: All right! With cream and two sugars. Now get on the phone.
(Condi picks up the phone.)
Condi: Rice, here.
George: Rice? Good idea. And a couple of egg rolls, too. Maybe we should
send some to the guy in China. And the Middle East. Can you get Chinese
food in the Middle East?

 

---

RESERVATIONS OF AN AIRLINE AGENT

(After Surviving 130,000 Calls from the Traveling Public)
By: Jonathan Lee-The Washington Post

I work in a central reservation office of an airline company. After more than 130,000 conversations-all ending with "Have a nice day and thanks for calling" -- I think it's fair to say that I'm a survivor.

I've made it through all the calls from adults who didn't know the difference between a.m. and p.m., from mothers of military recruits who didn't trust their little soldiers to get it right, from the woman who called to get advice on how to handle her teenage daughter, from the man who wanted to ride inside the kennel with his dog so he wouldn't have to pay for a seat, from the woman who wanted to know why she had to changeclothes on our flight between Chicago and Washington (she was told she'dhave to make a change between the two cities) and from the man who askedif I'd like to discuss the existential humanism that emanates from the soulof Habeeb.

In five years, I've received more than a boot camp education regarding the astonishing lack of awareness of our American citizenry. This lack of awareness encompasses every region of the country, economic status, ethnic background, and level of education. My battles haveincluded everything from a man not knowing how to spell the name of the townhe was from, to another not recognizing the name of "Iowa" as being a state,to another who thought he had to apply for a foreign passport to fly toWest Virginia. They are the enemy and they are everywhere.

In the history of the world there has never been asmuch communication and new things to learn as today. Yet, after askinga woman from New York what city she wanted to go to in Arizona, she asked "Oh... is it a big place?"

I talked to a woman in Denver who had never heard of Cincinnati, a man in Minneapolis who didn't know there was more than one cityin the South ("wherever the South is"), a woman in Nashville who asked, "Insteadof paying for my ticket, can I just donate the money to the National CancerSociety?" and a man in Dallas who tried to pay for his ticket by sticking quarters in the pay phone he was calling from.

I knew a full invasion was on the way when, shortlyafter signing on, a man asked if we flew to exit 35 on the New Jersey Turnpike. Then a woman asked if we flew to area code 304. And I knew I hadbeen shipped off to the front when I was asked, "When an airplane comesin, does that mean it's arriving or departing?" I remembered the stricttraining we had received -- four weeks of regimented classes on airlinecodes, computer technology, and telephone behavior - and it allowed forno means of retaliation. We were told, "it's real hell out thereand ya got no defense." You're going to hear things so silly youcan't even make 'em up. You'll try to explain things to your friendsthat you don't even believe yourself, and just when you think you've heardit all, someone will ask if they can get a free round-trip ticket to Europeby reciting "Mary Had a Little Lamb."

It wasn't long before I suffered a direct hit from a woman who wanted to fly to Hippopotamus, NY. After assuring her that there was no such city, she became irate and said it was a big city with a big airport. I asked if Hippopotamus was near Albany or Syracuse. It wasn't. Then I asked if it was near Buffalo. "Buffalo!" she said. "I knew it was a big animal!"

Then I crawled out of my bunker long enough to be confronted by a man who tried to catch our flight in Maconga. I told him I'dnever heard of Maconga and we certainly didn't fly to it. But heinsisted we did and to prove it he showed me his ticket: Macon, GA.

I've done nothing during my conversational confrontationsto indicate that I couldn't understand English. But after quotingthe round-trip fare the passenger just asked for, he'll always ask: "...Is that one-way?" I never understood why they always question if whatI just gave them is what they just asked for.

But I've survived to direct the lost, correct the wrong, comfort the weary, teach U.S. geography and give tutoring in the spelling and pronunciation of American cities. I have been told things like: "I can't go stand-by for your flight because I'm in a wheelchair." I've been asked such questions as: "I have a connecting flight to Knoxville. Does thatmean the plane sticks to something?" And once a man wanted to go toIllinois. When I asked what city he wanted to go to in Illinois,he said, "Cleveland, Ohio."

After 130,000 little wars of varying degrees, I'm a wise old veteran of the communication conflict and can anticipate with accuracywhat the next move by "them" will be. Seventy-five percent won't haveanything to write on. Half will not have thought about when they're returning. A third won't know where they're going; 10 percent won't care where they're going. A few won't care if they get back. And James will bethe first name of half the men who call.

But even if James doesn't care if he gets to the city he never heard of; even if he thinks he has to change clothes on our plane that may stick to something; even if he can't spell, pronounce, or remember whatcity he's returning to, he'll get there because I've worked very hard tomake sure that he can. Then with a click of the phone, he'll becomea part of my past and I'll be hoping the next caller at least knows whatday it is.
Oh, and James... "Thanks for calling and have a nice day."

---

LINES OF QUESTIONING WORTH QUESTIONING

Recently reported in the Massachusetts Bar Association Lawyers Journal, the following are questions actually asked of witnesses by attorneys during trials and, in certain cases, the responses given byinsightful witnesses:

1. "Now doctor, isn't it true that when a person dies in his sleep, he doesn't know about it until the next morning?"

2. "The youngest son, the twenty-year old, how old is he?"

3. "Were you present when your picture was taken?"

4. Q: "Doctor, before you performed the autopsy, did you check for a
pulse?"
A: "No."
Q: "Did you check for blood pressure?"
A: "No."
Q: "Did you check for breathing?"
A: "No."
Q: "So, then it is possible that the patient was alive when you began
the autopsy?"
A: "No."
Q: "How can you be so sure, Doctor?"
A: "Because his brain was sitting on my desk in a jar."
Q: "But could the patient have still been alive nevertheless?"
A: "It is possible that he could have beenalive and practicing law
somewhere."

5. "Was it you or your younger brother who was killed in the war?"

6. "Did he kill you?"

7. "How far apart were the vehicles at the time of the collision?"

8. "You were there until the time you left, is that true?"

9. "How many times have you committed suicide?"

10. Q: "So the date of conception (of the baby) was August 8th?"
A: "Yes."
Q: "And what were you doing at that time?"

11. Q: "She had three children, right?"
A: "Yes."
Q: "How many were boys?"
A: "None."
Q: "Were there any girls?"

12. Q: "You say the stairs went down to the basement?"
A: "Yes."
Q: "And these stairs, did they go up also?"

13. Q: "Mr. Slatery, you went on a rather elaborate honeymoon, didn't
you?"
A: "I went to Europe, Sir."
Q: "And you took your new wife?"

14. Q: "How was your first marriage terminated?"
A: "By death."
Q: "And by who's death was it terminated?"

15. Q: "Can you describe the individual?"
A: "He was about medium height and had a beard."
Q: "Was this a male, or a female?"

16. Q: "Is your appearance here this morning pursuant to a deposition
notice which I sent to your attorney?"
A: "No this is how I dress when I go to work."

17. Q: "Doctor, how many autopsies have you performed on dead people?"
A: "All my autopsies areperformed on dead people."

18. Q: "All your responses must be oral, OK? What school did you goto?"
A: "Oral."

19. Q: "Do you recall the time that you examined the body?"
A: "The autopsy started around 8:30 p.m.."
Q: "And Mr. Dennington was dead at the time?"
A: "No, he was sitting on the table wondering why I was doing
anautopsy."

20. Q: "You were not shot in the fracas?"
A: "No, I was shot midway between the fracas and the navel."

21. Q: "Are you qualified to give a urine sample?"
A: "I have been since early childhood."

 

---

About a century or two ago, the Pope decided that all the Jews had to leave the Vatican. Naturally there was a big uproar from the Jewish community. So the Pope made a deal. He would have a religious debate with a member of the Jewish community. If the Jew won, theJews could stay. If the Pope won, the Jews would leave.

The Jews realized that they had no choice. So they picked a middle-aged man named Moishe to represent them. Moishe asked for one additionto the debate. To make it more interesting, neither side would be allowedto talk. The pope agreed.

The day of the great debate came. Moishe and the Pope sat opposite each other for a full minute before the Pope raised his hand and showed three fingers. Moishe looked back at him and raised one finger. The Pope then waved his fingers in a circle around his head. Moishe pointedto the ground where he sat. The Pope then pulled out a wafer and a glassof wine. Moishe then pulled out an apple. The Pope stood up and said, "Igive up. This man is Too Good. The Jews can Stay."

An hour later, the cardinals were all around the Pope asking him what happened. The Pope said, "First I held up 3 fingers to represent the Trinity. He responded by holding up one finger to remind me that there was still one G-d common to both our religions. Then I waved my finger around me to sho him that G-d was all around us. He responded by pointing to the ground and showing that G-d was also right here with us. Ithen pulled out the wine and the wafer to show that G-d absolves us fromour sins. He pulled out an apple to remind me of original sin. He had an answer for everything. What could I do?"

Meanwhile, the Jewish community had crowded around Moishe. "What Happened?" they all asked. "Well," said Moishe. "First he said to methat the Jews had three days to get out of here. I told him that notone of us was leaving. Then he told me that the whole city would becleared of Jews. I let him know that we were staying right here."

"And then?" asked a woman.

"I don't know," said Moishe. "He then took out his lunch and I took out mine."

 

---

The Top 15 Signs Your Co-Workers Jewish Holiday Wasn't Strictly Observed.....

15. Telltale mouse ears on his new "Yarmulke".

14. Sun worshipping appears to be the only thing accomplishedthat was even remotely religious.

13. Thinks that the "Torah" is something you wear to a frat party.

12. Comes back wearing a "Club Med: The Sea may be Dead, but the Night-life isn't."

11. You happen to know there are no High Holy Day services atSanta Anita Racetrack.

10. Kareem in Accounting keeps calling it "Yom Shakur."

9. As far as you know, circumcisions don't "grow back".

8. Doesn't know the difference between Hebrew and Home-brew.

7. She's complaining that Kathie Lee isn't really on all Carnival Cruises.

6. Claims he was observing "Chaka Khan."

5. Menorah on his desk displays 3 sleeves worth of golf balls.

4. Thinks "Rosh Hashanah" is a song by "The Knack".

3. His yarmulke has two cans of beer and a drinking straw.

2. "And if the rabbi sees his shadow when he comes out of temple, there'll be 4 more months of summer."

And the #1 Sign Your Co-Workers Jewish Holiday Wasn't Strictly Observed....

1. Took off all of LAST month for Ramadan.

 

---

Boris, Bill and Bibi have good news and bad news from God.

God, having decided to destroy the world, reveals His intention to 3 world leaders:
Bill Clinton
Boris Yeltsin and
Bibi Netanyahu.

Boris Yeltsin, in an address to the Russian people, said: "I have bad news and more bad news. First of all, in contrast to what wehave been taught to believe, God does exist. He appeared to me andspoke to me. Secondly, He intends to destroy the world."

At the same time, in an address to the American People, Bill Clinton said: "There's good news and bad news. First of all, as we have been taught to believe, God exists; he appeared to me and spoke to me. The bad news is that He intends to destroy the world."

At the same time, in an address to the Israeli people, Bibi Netanyahu said, "There's good news and more good news. First of all, as we havebeen taught to believe, God exists; He appeared to me and spoke tome. Furthermore, based on what He said to me, I can assure you that a Palestinian state will NOT be established."

 

---

Team Sports...

Yeshiva University decided to field a crew team. Unfortunaely, they lost race after race. They practiced for hours every day, but never managed to come in any better than dead last.

The Rosh Yeshiva finally decided to send Yankel to spy on the Harvard team. So Yankel shlepped off to Cambridge and hid in the bullrushes off the Charles River, from where he carefully watched the Harvard teamas they practiced.

Yankel finally returned to Yeshiva. "I have figured out theirsecret," he announced. "They have eight guys ROWING and only ONE guy shouting."

Quick Retort

A man boards a plane with 6 kids.

After they get settled in their seats a woman sitting across the
aisle leans over to the man and asks, 'Are all of these kids
yours?'

He replies, 'No. I work for a condom company. These are customer
complaints.'

Wish I could think so quickly...

---

HOW TO SAVE THE AIRLINES

Dump the male flight attendants. No one wanted them in the first place.

Replace all the female flight attendants with good-looking strippers! What the hell--They don't even serve food anymore, so what's the loss?

The strippers would at least triple the alcohol sales and get a 'party atmosphere' going in the cabin. And, of course, every businessman in this country would start flying again, hoping to see naked women.

Because of the tips, female flight attendants wouldn't need a salary, thus saving even more money.  I suspect tips would be so good that we could charge the women for working the plane and have them kick back 20% of the tips, including lap dances and 'special services.'

Hijackings would come to a screeching halt, and the airline industry would see record revenues.

This is definitely a win-win situation if we handle it right--a golden opportunity to turn a liability into an asset.

Why didn't Bush think of this? Why do I still have to do everything myself?

Sincerely,
Bill Clinton

---

Theological Golf

O'Brian loved to play golf and would go out alone to a course andget paired up with any group that needed a fourth. One day he went to hisfavorite course and the starter said, "I'm sorry O'Brian, but theonly group I can put you with is one with three Chassidic rabbis."

O'Brian says, "That's fine with me."

He joins the group and tees off. His shot is about 200 yards out and off to the right rough. Reb Moshe tees off 300 yards straight out into the middle of the fairway. Reb Yitzchak's shot is about 290 and Reb Yaacov's is 300, but slightly off center. O'Brian has trouble with getting out of the rough and three-putts, while the rabbis' approach shots are right on the pin, they two-putt for par.

The rest of the round is the same, with the rabbi's scores eitherpar or under par, while O'Brian has a 90. He says to them, "You guys mustplay and practice all the time."

Reb Yitzchak says, "No, we study all the time and only play once a week. But, on our Sabbath, while we are in shul, we say a prayer asking God to give us one good round of golf each week."

O'Brian is so impressed that he goes home and tells his wife thatthey are converting. They study, convert, join a shul, and go to servicesevery Shabbat.

About a year later, O'Brian runs into the threesome at the same course and they invite him to play with them. The game is exactly like last year's. O'Brian is doing nothing right, and the three are perfect. At the end,O'Brian says to the rabbis, "I don't understand it. I converted, joined a shul, pray every week."

Reb Moshe says, "You joined a shul? Which one?"

O'Brian says, "Beth El."

Reb Moshe says, "No, no, no! Beth El is for TENNIS!"

---

Two Bees

Two bees buzz around what's left of a rose bush. "How was your summer?" asks bee number one. "Not too good," says bee two. "Lotta rain, lotta cold. Not enough flowers, not enough pollen." The first bee has an idea. "Hey, why don't you go down the corner and hanga left? There's a bar mitzvah going on. Plenty of flowers and fruit." Beetwo buzzes, "Thanks!" and takes off. An hour later, the bees bump into eachother again. "How was the bar mitzvah?" asks the first bee. "Great!" saysthe second. The first bee peers at his pal and wonders, "What's that on your head?" "A yarmulke," the bee answers. "I didn't want them tothink I was a wasp."

---

An American couple arrived on holiday in Jerusalem, and passing by the Western Wall noticed, apart from the generality of those praying, one man in particular, whose devotions seemed even more intense than the others, to the extent that from time to time he would beat his hands, and even his forehead, against it. They continued with their sight-seeing, and returning to their hotel for lunch they noticed that while most had gone, he was still there. In the afternoon their route took them in that direction again, and there he was, and similarly on theirreturn at tea time, although it was obvious that all the other personnelhad changed on each occasion.

Very intrigued, they walked back again after dinner,and there he was, but as they watched he finished and began to fold uphis tallis, so they walked over to talk to him. "Forgive us for intruding,but we have passed this way five times today, and on each occasionwe have noticed that you were here, praying. Everyone else seems tostay for an hour or so, but it appears to us that you have been hereall day. Can you explain to us, what special prayers are you sayingthat need so much time?"

"Well," he replied, "it is true that I am here all day every day, but I pray according to my own system. For the first two hours or so I pray for myself, my wife, my children, the whole of my family. For the next two hours I pray for the State of Israel, that it should prosper in peace. Then I pray for the whole of the Middle East, that there should be understanding, friendship with our neighbours, and no more wars or terrorism. Finally I pray for the whole world, for an end to poverty, pain, suffering, and animosity, and for universal peace." The visitors were most impressed. Tears came to their eyes. "That is fantastic, such piety, such nobility of character,such selflessness! That anyone could be so devout; what must it belike, what must it feel like?"

"Ah, what does it feel like?

You want to know what it feels like?? It feels like I'm talking to a wall!!"

---

Headline: MAJOR TECHNOLOGICAL BREAKTHROUGH

Introducing the new Bio-Optic Organized Knowledge device, trade named BOOK.

BOOK is a revolutionary breakthrough in technology: no wires, no electric circuits, no batteries, nothing to be connected or switched on. It's soeasy to use, even a child can operate it.

Compact and portable, it can be used anywhere -- even sitting in an armchair by the fire -- yet it is powerful enough to hold as much information as a CD-ROM disc. Here's how it works:

BOOK is constructed of sequentially numbered sheets of paper (recyclable), each capable of holding thousands of bits of information.

The pages are locked together with a custom-fit device called a binder which keeps the sheets in their correct sequence. Opaque Paper Technology (OPT) allows manufacturers to use both sides of the sheet, doubling theinformation density and cutting costs. Experts are divided on the prospectsfor further increases in information density; for now, BOOKS with more informationsimply use more pages.

Each sheet is scanned optically, registering information directlyinto your brain. A flick of the finger takes you to the next sheet.

BOOK may be taken up at any time and used merely by opening it.

BOOK never crashes or requires rebooting, though like other display devices it can become unusable if dropped overboard. The "browse" feature allows you to move instantly to any sheet, and move forward or backwardas you wish. Many come with an "index" feature, which pin-points the exactlocation of any selected information for instant retrieval.

An optional "BOOKmark" accessory allows you to open BOOK to the exact place you left it in a previous session -- even if the BOOK has been closed. BOOKmarks fit universal design standards; thus, a single BOOKmark can be used in BOOKs by various manufacturers. Conversely, numerous BOOK markers can be used in a single BOOK if the user wants to store numerous views at once. The number is limited only by the number of pages in the BOOK.

You can also make personal notes next to BOOK text entries with an optional programming tool, the Portable Erasable Nib Cryptic Intercommunication
Language Stylus (PENCILS).

Portable, durable, and affordable, BOOK is being hailed as a precursor of a new entertainment wave. Also, BOOK's appeal seems so certain that thousands of content creators have committed to the platform and investors are reportedly flocking. Look for a flood of new titles soon.

---

The Lone Ranger and Tonto walked into a bar one day and sat down to drink a beer. After a few minutes, a big tall cowboy walked in and and said "Who owns the big white horse outside." The Lone Ranger stoodup, hitched his gunbelt, and said, "I do , why?" The cowboy looked at theLone Ranger and said "I just thought you'd like to know that your horseis about dead outside!!"

The Lone Ranger and Tonto rushed outside and, sure enough, Silverwas about dead from heat exhaustion. The Lone Ranger got him somewater and made him drink it and soon Silver was starting to feel a littlebetter. The Lone Ranger turned to Tonto and said, "Tonto, I want you torun around Silver and see if you can create enough of a breeze to makehim start to feel better.

"Tonto said, "Sure Kemosabe" and took off running circles around Silver. Not able to do anything else but wait, the Lone Ranger returned to thebar to finish his drink.

A few minutes later, another cowboy struts into the bar and announces, "Who owns that big white horse outside?" The Lone Ranger stands again,and claims, "I do, what's wrong with him this time?" The cowboy says tohim, "Nothing much, I just wanted you to know...........................

You left your Injun runnin"!!!"

---

George Carlin's warped thoughts!

1. If a mute swears, does his mother wash his hands with soap?
2. If someone with multiple personalities threatens to kill himself, is it
considered a hostage situation?
3. Instead of talking to your plants, if you yelled a them would they still
grow, only to be troubled and insecure?
4. What's another word for synonym?
5. Isn't it a bit unnerving that doctors call what they do practice"?
6. When sign makers go on strike, is anything written on their signs?
7. When you open a bag of cotton balls, is the top one meant to be thrown
away?
8. Where do forest rangers go to "get away from it all"?
9. Why isn't there mouse-flavored cat food?
10. Why do they report power outages on TV?
11. What should you do when you see an endangered animal that is eating an
endangered plant?
12. Is it possible to be totally partial?
13. If a parsley farmer is sued, can they garnish his wages?
14. Would a fly that loses it wings be called a walk?
15. Why do they lock gas station bathrooms? Are they afraid someone will
clean them?
16. If the funeral procession is at night, do folks drive with their
headlights off?
17. If a stealth bomber crashes in a forest, will it make a sound?
18. If a man speaks in the forest and there is no woman to hear him, is he
still wrong?
19. If a turtle loses his shell, is it naked or homeless?
20. Why don't sheep shrink when it rains?
21. Should vegetarians eat animal crackers?
22. If the cops arrest a mime, do they tell him he has the right to remain
silent?
23. Why do people who know the least know it the loudest?
24. If vegetarians eat vegetables, what do humanitarians eat?

---

The madame opened the brothel door to see an elderly Jewish man. His clothes were all disheveled and he looked 'needy. "Can I help you?" the madame asked. "I want Natalie," the old man replied. "Sir, Natalie is one of our most expensive ladies perhaps someone else..." "No, I want Natalie." Just then Natalie appeared and announced to the old man that she charged $1,000 per hour. The man never blinked and reached into his pocket and handed her ten $100 bills. The two went up to a room for an hour whereupon the mancalmly left. The next night he appeared again demanding Natalie. Natalie explained that no one had ever come back two nights in a row and that there were no discounts... it was still $1,000 for one hour. Again the old man took out the money, the two went up to the room and he calmly left an hour later. When he showed up the third consecutive night no one could believe it.

Again he handed Natalie the money and up to the room they went. At the end of the hour Natalie questioned the old man: "No one has ever used my services three nights in a row...where are you from?" The old man replied, "I am from Minsk." "Really", replied Natalie, "I have a sister who livesthere." "I know," said the old man, "she gave me $3,000 to give to you."

---

Howard had felt guilty all day long. No matter how much he tried to forget about it, he couldn't. The guilt and sense of betrayal was overwhelming.

But, every once in a while, he'd hear that soothing voice trying to reassure him -- "Howard. Don't worry about it. You aren't the first doctor to sleep with one of his patients, and you won't be the last." But, invariably, the other voice would bring him back to reality --

"Howard. You're a veterinarian."

---

MOMMY'S LITTLE GIRL

A little girl and her mother were out and about. The girl, out ofthe blue, asked her mother, "Mommy, How old are you?"

The mother responded, "Honey, women don't talk about their age. You'll learn this as you get older.

The girl then asked, "Mommy, how much do you weigh?"

Her mother responded again, "That's another thing women don't talk about. You'll learn this, too, as you grow up."

The girl still wanting to know about her mother, then fires off another question, "Mommy, Why did you and Daddy get a divorce?"

The mother, a little annoyed by the questions, responded, "Honey,that is a subject that hurts me very much, and I don't want to talk aboutit now."

The little girl, frustrated, sulks until she is dropped off at a friend's house to play. She consults with her girlfriend about her and her mother's conversation. The girlfriend says, "All you have to do is sneak and look at your mother's driver's license. It's just a like a report card from school. It tells you everything." Later, the little girl and her mother are outand about again. The little girl starts off with, "Mommy, Mommy, I know how old you are. I k now how old you are. You're 32 years old."

The mother is very shocked. She asks, "Sweetheart, how do you know that?"

The little girl shrugs and says, "I just know. And I know how much you weigh. You weigh 130 pounds."

"Where did you learn that?"

The little girl says, "I just know. And I know why you and daddy got a divorce. You got an "F" in sex."

---

SEMINARS FOR FEMALES
(prepared and presented by males)
----------------------------------------
OK, Here's the Female version.

1. Are You Ready to Leave?: Definition of the Word YES
2. Appropriate Rhetorical Questions (formerly titled "Honey, Do ILook Fat?")
3. Elementary Map Reading
4. Crying and Law Enforcement
5. Advanced Math Seminar: Programming Your VCR
6. You CAN Go Shopping for Less than 4 Hours
7. Gaining Five Pounds vs. The End of the World: A Study in Contrast
8. The Seven-Outfit Week
9. PMS: It's YOUR Problem, Not Mine (formerly "It's Happened Monthly Since Puberty: Deal With It")
10. Driving : Makeup and Driving: It's As Simple As Oil and Water
11. Beyond "Clean and Dirty": The Nuances of Wearable Laundry
12. Joys of the Remote Control: Reaping the Benefits of 50+ Channels
13. We Forget Birthdays, You Forget Sports Stats: LET'S LET IT DROP
14. Know When to Say When: The Limits of Makeup
15. Telephone Translations (formerly titled "Me Too Equals I LoveYou")
16. Yes, You Too Can Buy Condoms (formerly titled "WE learned to deal with the embarrassment")
17. Gift-giving Fundamentals (formerly titled "Fabric Bad, Electronics Good")
18. Putting the Seat Down By Yourself: Potential Energy is on Your Side
19. MYOB: Proper Response to Other Couple's Public Arguments
20. The Penis: His Best Friend Can Be Yours Too
21. Your Mate: Selfish Bastard, or Victimized Sensitive Man?

---

Each simile listed below was actually used by high school students in their various essays and short stories:

He spoke with the wisdom that can only come from experience, like a guy who went blind because he looked at a solar eclipse without oneof those boxes with a pinhole in it and now goes around the country speaking at high schools about the dangers of looking at a solar eclipse withoutone of those boxes with a pinhole in it.

She caught your eye like one of those pointy hook latches that used to dangle from screen doors and would fly up whenever you banged the door open again.

The little boat gently drifted across the pond exactly the way a bowling ball
wouldn't.

McBride fell 12 stories, hitting the pavement like a Hefty bag filled with vegetable soup.

From the attic came an unearthly howl. The whole scene had an eerie, surreal quality, like when you're on vacation in another city and "Jeopardy" comes on at 7 P.M. instead of 7:30.

Her hair glistened in the rain like nose hair after a sneeze.

Her eyes were like two brown circles with big black dots inthe center.

Her vocabulary was as bad as, like, whatever.

He was as tall as a six-foot, three-inch tree.

The hailstones leaped from the pavement, just like maggots when you fry them in hot grease.

Her date was pleasant enough, but she knew that if her lifewas a movie, this guy would be buried in the credits as something like"second tall man".

Long separated by cruel fate, the star-crossed lovers race across the grassy field toward each other like two freight trains, onehaving left Cleveland at 6:36 P.M. traveling at 55 mph, the other fromTopeka at 4:19P.M. at a speed of 35 mph.

The politician was gone but unnoticed, like the period after the Dr. on a Dr. Pepper can.

They lived in a typical suburban neighborhood with picket fences that resembled Nancy Kerrigan's teeth.

John and Mary had never met. They were like two hummingbirds who had also never met.

The thunder was ominous-sounding, much like the sound of a thin sheet of metal being shaken backstage during the storm scene in a play.

The red brick wall was the color of a brick-red Crayola crayon.

His thoughts tumbled in his head, making and breaking alliances like underpants in a dryer without Cling Free.

---

The World's Shortest Books

25. "My Plan To Find The Real Killers" by OJ Simpson
24. "To All The Men I've Loved Before" by Ellen DeGeneres
23. "The Book of Virtues" by Dean Rains
22. The Difference between Reality and Dilbert
21. Human Rights Advances in China
20. "Things I Wouldn't Do for Money" by Dennis Rodman
19. Al Gore: The Wild Years
18. Amelia Earhart's Guide to the Pacific Ocean
17. America's Most Popular Lawyers
16. Career Opportunities for Liberal Arts Majors
15. Detroit - A Travel Guide
14. Different Ways to Spell "Bob"
13. Dr. Kevorkian's Collection of Motivational Speeches
12. Easy UNIX
11. Ethiopian Tips on World Dominance
10. Everything Men Know About Women
9. Everything Women Know About Men
8. French Hospitality
7. George Foreman's Big Book of Baby Names
6. "How to Sustain a Musical Career" by Art Garfunkel
5. Mike Tyson's Guide to Dating Etiquette
4. One Hundred and One Spotted Owl Recipes by the EPA
3. Staple Your Way to Success
2. The Amish Phone Directory
1. The Engineer's Guide to Fashion

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THINGS YOU WOULD NEVER KNOW WITHOUT THE MOVIES

During all police investigations it will be necessary to visit a strip club at least once.

All telephone numbers in America begin with the digits 555.

All beds have special L-shaped cover sheets which reach up to thearmpit level on a woman but only to waist level on the man lying besideher.

The ventilation system of any building is the perfect hiding place. No-one will ever think of looking for you in there and you can travel to any other part of the building you want without difficulty.

Should you wish to pass yourself off as a German officer, it willnot be necessary to speak the language. A German accent will do.

A man will show no pain while taking the most ferocious beating but will wince when a woman tries to clean his wounds.

Kitchens don't have light switches. When entering a kitchen at night, you should open the fridge door and use that light instead.

If staying in a haunted house, women should investigate any strange noises in their most revealing underwear.

Cars that crash will almost always burst into flames.

Wearing a vest or stripping to the waist can make a man invulnerable to bullets.

If you find yourself caught up in a misunderstanding that could be cleared up quickly with a simple explanation, for goodness sake, keep your mouth shut.

Any person waking from a nightmare will sit bolt upright and pant.

A cough is usually the sign of a terminal illness.

All bombs are fitted with electronic timing devices with large red readouts so you know exactly when they're going to go off.

When in love, it is customary to burst into song.

When confronted by an evil international terrorist, sarcasm and wisecracks are your best weapons.

One man shooting at 20 men has a better chance of killing them than 20 men firing at 1 man.

Creepy music coming from a cemetery should always be investigatedmore closely.

If being fired at by Germans, hide in a river - or even a bath. German bullets are unable to penetrate water.

Most laptop computers are powerful enough to override the communication systems of any invading alien civilization.

Freelance helicopter pilots are always eager to accept bookings from international terrorist organizations - even though the job will require them to shoot total strangers and will end in their own certain death as thehelicopter explodes in a ball of flames.

Most people keep a scrapbook of newspaper clippings - especially if any of their family or friends have died in a strange boating accident.

All computer disks will work in all computers, regardless of software.

Police Departments give their officers personality tests to make sure they are deliberately assigned a partner who is their total opposite.

When they are alone, all foreigners prefer to speak English to each other.

Action heroes never face charges for manslaughter or criminal damage despite laying entire cities to waste by their actions.

You can always find a chainsaw when you need one. Any lock can bepicked by a credit card or a paper clip in seconds - unless it's the doorto a burning building with a child trapped inside.

You can tell if somebody is British because they will be wearing a bow tie.

When driving a car it is normal to look not at the road but at the person sitting beside you or in the back seat for the entire journey.

An electric fence, powerful enough to kill a dinosaur will cause no lasting damage to an eight year old child.

Having a job of any kind will make father's forget their son's eighth birthday.

Honest and hard working policemen are traditionally gunned down three days before their retirement.

If you are blonde and pretty, it is possible to become a world expert in Nuclear Fission at age 22.

The more a man and a woman hate each other, the more likely they will fall in love.

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Lawyers

A small town prosecuting attorney called his first witness to thestand in a trial--a grandmotherly, elderly woman. He approached her andasked, "Mrs. Jones, do you know me?" She responded, "Why, yes, I do knowyou Mr. Williams. I've known you since you were a young boy. And frankly,you've been a big disappointment to me. You lie, you cheat on your wife,you manipulate people and talk about them behind their backs. You thinkyou're a rising big shot when you haven't the brains to realize you neverwill amount to anything more than a two-bit paper pusher. Yes, I know you."The lawyer was stunned. Not knowing what else to do he pointed across theroom and asked, "Mrs. Williams, do you know the defense attorney?" She againreplied, "Why, yes I do. I've known Mr. Bradley since he was a youngster,too. I used to babysit him for his parents. And he, too, has been a realdisappointment to me. He's lazy, bigoted, he has a drinking problem. Theman can't build a normal relationship with anyone and his law practice isone of the shoddiest in the entire state. Yes, I know him." At this point,the judge rapped the courtroom to silence and called both counselors to thebench. In a very quiet voice, he said with menace, "If either of you asksher if she knows me, you'll be in jail for contempt within 5 minutes!"

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THE PERFECT DAY - HER
8:45 Wake up to hugs and kisses
9:00 5 lbs lighter on the scale
9:30 Light Breakfast
11:00 Sunbathe
12:30 Lunch with best friend at outdoor cafe
1:45 Shopping
2:30 Run into boyfriend's/husband's ex - notice she's gained 30lbs
3:00 Facial, massage, nap
7:30 Candlelight dinner for two and dancing
10:00 Make love
11:30 Pillow talk in his big strong arms

THE PERFECT DAY - HIM
6:45 Alarm.
7:00- 7:30 Shower and massage.
7:30- 7:45 Blowjob.
7:45- 8:15 Massive dump while reading USA Today sports section.
8:15 Limo arrives, Stoli Bloody Marys.
8:30 Butler Aviation, O'Hare Field, Lear Jet to Augusta, Georgia.
9:30 Front nine holes, Augusta National Golf Club.
11:30-12:30 Lunch - 2 dozen oysters, 3 Heinekens.
12:30-12:45 Blowjob.
12:45- 2:30 Back nine holes, Augusta National Golf Club.
2:30 Limo to Augusta Airport, Bombay Sapphire Martini.
3:30- 6:15 Nassau, Bahamas, Afternoon fishing with all female crew (topless). Catch 1249 lb. Blue Marlin. Grilled tuna and steamed lobster appetizers, six Heinekens, nap.
6:15- 6:30 Blowjob.
6:30 Lear Jet return flight, total body massage in transit.
7:30 Shit, shower, shave.
8:00- 9:00 Watch CNN Live coverage of Bill Clinton's resignation. Hillary and Al Gore are indicted in the same scandal (which involves graphic pictures and large farm animals).
9:00-10:30 Dinner at Ritz Carlton, Oysters Casino, 20oz. Filet mignons (rare), Gorgonzola salad, Fettucini Alfredo, ChateauLafite Rothschild 1963 (magnum) creme brule, Louis XII Cognac, Cohiba Lancero
10:30-11:30 Sex with 3 women (minimum two with mixed racialorigin)
11:30-12:00 Whirlpool, steam and massage. Women quietly get dressed, hail cab and leave.
Midnight Sleep

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Men

Men like to barbecue. Men will cook if danger is involved.

If you buy your husband or boyfriend a video camera, for the first few weeks he has it, lock the door when you go to the bathroom. Mostof my husband's early films ended with a scream and a flush.

Men are very confident people. My husband is so confident that when he watches sports on television, he thinks that if he concentrates he can help his team. If the team is in trouble, he coaches the players from our living room, and if they're really in trouble, I have to getoff the phone in case they call him.

Men love to be the first to read the newspaper in the morning. Not being the first is upsetting to their psyches.

All men look nerdy in black socks and sandals.

Don't try to teach men how to do anything in public. They can learn in private; in public they have to know.

All men are afraid of eyelash curlers. I sleep with one under my pillow, instead of a gun.

A good place to meet a man is at the dry cleaner. These men usually have jobs and bathe.

Men love watches with multiple functions. My husband has one that isa combination address book, telescope and piano.

Men are sensitive in strange ways. If a man has built a fire and thelast log does not burn, he will take it personally.

Men are brave enough to go to war, but they are not brave enough to get a bikini wax.

Men don't get cellulite. God might just be a man.

Men have an easier time buying bathing suits. Women have two types: depressing and more depressing. Men have two types: nerdy and not nerdy.

Men have higher body temperatures than women. If your heating goes out in winter, I recommend sleeping next to a man. Men are like portable heaters that snore.

Most men hate to shop. That's why the men's department is usually onthe first floor of a department store, two inches from thedoor.

If a man prepares dinner for you & the salad contains three or more types of lettuce, he's serious.

Men own basketball teams. Every year cheerleaders' outfits get tighter and briefer, and players' shorts get baggier and longer.

When four or more men get together, they talk about sports. When four or more women get together, they talk about men.

Not one man in a beer commercial has a beer belly.

Men are less sentimental than women. No man has ever seen the movie 'The Way We Were' twice, voluntarily.

Most women are introspective: "Am I in love? Am I emotionally and creatively fulfilled?" Most men are outrospective: "Did my team win? How's my car?"

Men hate to lose. I once beat my husband at tennis. I asked him, "Are we going to have sex again?" He said, "Yes, but not with each other."

Getting rid of a man without hurting his masculinity is a problem. "Get out" and "I never want to see you again" might sound like a challenge. If you want to get rid of a man, I suggest saying, "I love you... I want to marry you... I want to have your children." Sometimes they leave skid marks. [I recognize this as being from Rita Rudner; maybe the rest are, too -- IB]

Men who listen to classical music tend not to spit.

Only men who have worn a ski suit understand how complicated it is for a woman to go to the bathroom when she's wearing a jumpsuit.

When a woman tries on clothing from her closet that feels tight, shewill assume she has gained weight. When a man tries something from hiscloset that feels tight, he will assume the clothing has shrunk.

Male menopause is a lot more fun than female menopause. With female menopause you gain weight and get hot flashes. Male menopause - you get to date young girls and drive motorcycles.

Men forget everything; women remember everything. That's why men need instant replays in sports. They've already forgotten what happened.

Men would like monogamy better if it sounded less like monotony.

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True Story

From Dick Pierce:
My son Nathaniel was heading home one night when he got pulled over by a local cop. Cop says: "I've been waiting all night for people like you." Nathaniel replies, "Well, I got
here as fast as I could." Cop told to get the hell out of there
and slow down.

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