一篇专门讲录音电平的短文
[SIZE=4][B]Level Practices in Digital Audio[/B][/SIZE]
This article is In Two Parts. PART 1 is called: The 20th Century---Dealing with the Peaks. OVERs, Levels, and Headroom, how to get the most from your equipment....
And Part 2 is called: How To Make Better Recordings in the 21st Century---An Integrated Approach to Metering, Monitoring, and Leveling Practices
------------
[SIZE=4][B]PART I: The 20th Century---Dealing With The Peaks[/B][/SIZE]
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 DAT machine that says a tape peaks to -1 dB while another machine shows an OVER level, yet your workstation 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. An alternate version of this article appeared in the March issue of Mix Magazine.
[SIZE=4][B]Section I: Digital Meters and OVER Indicators[/B][/SIZE]
DAT recorder manufacturers pack a lot in a little box, often compromising on meter design to cut costs. A few 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). 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 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 (long before an OVER actually occurs). But even if the meter has a segment at every decibel, on playback, the machine can't tell the difference between a level of 0 dBFS (FS = Full Scale) and an OVER. Distinguishing between these two requires intelligence that I've never seen on a DAT machine or a typical DAW. 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 the number 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, the DAT machine's OVER indicator 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 DAT machine. I've already received several overloaded tapes which were traced to an external A/D converter that wasn't equipped with an overload indicator.
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 tape 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. However, if you have to rely on the built-in OVER indicator of a DAT machine, only experience with that machine will tell how accurate it is. With a DAT machine's meter, it may be better not to exceed -1 dB on music peaks. 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 tape 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.
[SIZE=4][B]Section II: How Loud is It?[/B][/SIZE]
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 sayin 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 digital tape, 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 DATs, 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-model 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.
[SIZE=4][B]Section III: Calibrating Studio Levels[/B][/SIZE]
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.
One of the most common mistakes made by digital equipment manufacturers is to assume that, if the digital signal "clips" at 0 dBFS, then it's OK to install a (cheap) analog output stage that would clip at a voltage equivalent to, say, 1 dB higher. This almost guarantees a nasty-sounding DAT recorder, because of the lack of cushion in its analog output section.
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. That's why some DAT machines have marks at -18 dB or 14 dB. I'd like to see accurate calibration marks on digital recorders 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. That's also why each manufacturer's DAT recorder has a different analog output level. It used to be easy to match a recorder to a console. Only one major manufacturer of DAT machines provides user calibration trims for analog inputs and outputs. My least favorite DAT machines have fixed output levels, and I've installed custom trimpots in many of them.
----In 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 digital tape, standardizing on -20 dBFS will simplify the process. Recording studios who decide on -18 dBFS ~0 VU (a standard used by a popular DAT manufacturer) 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 may be tolerated.
----Mastering Studios
Mastering studios are working more frequently in 20-bit or 24-bit. In Part II (below) 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.
Copyright Notice and Linking permission: These HTML documents are Copyright 1995-2002 Digital Domain, Inc. Except for Part II, which copyright may soon be assigned to the Audio Engineering Society. This article last revised October 2002. The following are trademarks of Digital Domain, Inc.: Digital Domain, Digi-nary. These documents may not be reproduced in any manner without the permission of the copyright owner. We invite the audio and music community to link to this web site, which will be periodically revised.
This article is In Two Parts. PART 1 is called: The 20th Century---Dealing with the Peaks. OVERs, Levels, and Headroom, how to get the most from your equipment....
And Part 2 is called: How To Make Better Recordings in the 21st Century---An Integrated Approach to Metering, Monitoring, and Leveling Practices
------------
[SIZE=4][B]PART I: The 20th Century---Dealing With The Peaks[/B][/SIZE]
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 DAT machine that says a tape peaks to -1 dB while another machine shows an OVER level, yet your workstation 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. An alternate version of this article appeared in the March issue of Mix Magazine.
[SIZE=4][B]Section I: Digital Meters and OVER Indicators[/B][/SIZE]
DAT recorder manufacturers pack a lot in a little box, often compromising on meter design to cut costs. A few 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). 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 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 (long before an OVER actually occurs). But even if the meter has a segment at every decibel, on playback, the machine can't tell the difference between a level of 0 dBFS (FS = Full Scale) and an OVER. Distinguishing between these two requires intelligence that I've never seen on a DAT machine or a typical DAW. 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 the number 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, the DAT machine's OVER indicator 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 DAT machine. I've already received several overloaded tapes which were traced to an external A/D converter that wasn't equipped with an overload indicator.
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 tape 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. However, if you have to rely on the built-in OVER indicator of a DAT machine, only experience with that machine will tell how accurate it is. With a DAT machine's meter, it may be better not to exceed -1 dB on music peaks. 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 tape 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.
[SIZE=4][B]Section II: How Loud is It?[/B][/SIZE]
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 sayin 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 digital tape, 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 DATs, 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-model 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.
[SIZE=4][B]Section III: Calibrating Studio Levels[/B][/SIZE]
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.
One of the most common mistakes made by digital equipment manufacturers is to assume that, if the digital signal "clips" at 0 dBFS, then it's OK to install a (cheap) analog output stage that would clip at a voltage equivalent to, say, 1 dB higher. This almost guarantees a nasty-sounding DAT recorder, because of the lack of cushion in its analog output section.
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. That's why some DAT machines have marks at -18 dB or 14 dB. I'd like to see accurate calibration marks on digital recorders 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. That's also why each manufacturer's DAT recorder has a different analog output level. It used to be easy to match a recorder to a console. Only one major manufacturer of DAT machines provides user calibration trims for analog inputs and outputs. My least favorite DAT machines have fixed output levels, and I've installed custom trimpots in many of them.
----In 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 digital tape, standardizing on -20 dBFS will simplify the process. Recording studios who decide on -18 dBFS ~0 VU (a standard used by a popular DAT manufacturer) 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 may be tolerated.
----Mastering Studios
Mastering studios are working more frequently in 20-bit or 24-bit. In Part II (below) 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.
Copyright Notice and Linking permission: These HTML documents are Copyright 1995-2002 Digital Domain, Inc. Except for Part II, which copyright may soon be assigned to the Audio Engineering Society. This article last revised October 2002. The following are trademarks of Digital Domain, Inc.: Digital Domain, Digi-nary. These documents may not be reproduced in any manner without the permission of the copyright owner. We invite the audio and music community to link to this web site, which will be periodically revised.
[SIZE=4][B]PART II:
How To Make Better Recordings in the 21st Century---An Integrated Approach to Metering, Monitoring, and Leveling Practices[/B][/SIZE]
updated from the article published in the September 2000 issue of the AES Journal
Bob Katz
[SIZE=4][B]A: Two-Channel[/B][/SIZE]
Introduction: 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 levelling 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 levelling, 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.
[SIZE=4][B]I. History: The VU Meter[/B][/SIZE]
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.
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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.
[SIZE=4][B]II. Current-day levelling problems[/B][/SIZE]
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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 . 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 meter lights 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?
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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.
[SIZE=4][B]III. The Magic of "83" with Film Mixes[/B][/SIZE]
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 4 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 ) 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 latter approach the source audio is uncompromised.
[SIZE=4][B]IV. The Magic of "-6 dB" Monitor Gain for the Home[/B][/SIZE]
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 11 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 over 100 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.
[SIZE=4][B]V. The relationship between SPL and 0 VU[/B][/SIZE]
Around 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 77 dB SPL with -20 dBFS pink noise).
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Since -20 dB FS 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.
[SIZE=4][B]VI. The K-System Proposal[/B][/SIZE]
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.
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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 dB FS analog reference points commonly used in Europe.
How To Make Better Recordings in the 21st Century---An Integrated Approach to Metering, Monitoring, and Leveling Practices[/B][/SIZE]
updated from the article published in the September 2000 issue of the AES Journal
Bob Katz
[SIZE=4][B]A: Two-Channel[/B][/SIZE]
Introduction: 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 levelling 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 levelling, 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.
[SIZE=4][B]I. History: The VU Meter[/B][/SIZE]
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.
[SIZE=4][B]II. Current-day levelling problems[/B][/SIZE]
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 . 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 meter lights 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.
[SIZE=4][B]III. The Magic of "83" with Film Mixes[/B][/SIZE]
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 4 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 ) 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 latter approach the source audio is uncompromised.
[SIZE=4][B]IV. The Magic of "-6 dB" Monitor Gain for the Home[/B][/SIZE]
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 11 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 over 100 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.
[SIZE=4][B]V. The relationship between SPL and 0 VU[/B][/SIZE]
Around 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 77 dB SPL with -20 dBFS pink noise).
Since -20 dB FS 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.
[SIZE=4][B]VI. The K-System Proposal[/B][/SIZE]
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 dB FS analog reference points commonly used in Europe.
[SIZE=4][B]VII. Production Techniques with the K-System[/B][/SIZE]
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 0 dB on the meter yields 83 dB SPL (per channel, C-Weighted, slow speed). 0 dB 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 0 dB (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 dB FS at 0 VU, and mix normally with the analog console and VUs. However, mixing consoles should be retrofitted 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 multitrack 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.
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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.
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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.
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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 computer. 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 the PC-compatible. 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, 3/11/01, 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 20-bit systems have 24 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 when 24-bit media reach the consumer. Until then chaos will remain in the classical field, and perhaps only metadata will sort out the classical music situation at the listener's end.
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.
[SIZE=4][B]VIII. An Extendable System[/B][/SIZE]
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.
[SIZE=4][B]IX. Metadata and the K-System[/B][/SIZE]
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-Systemfor 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.
[SIZE=4][B]X. In Conclusion[/B][/SIZE]
Let's bring audio into the 21st century. The K-system is the first integrated approach to monitoring, levelling practices, metering and metadata.
[SIZE=4][B]B: Multichannel[/B][/SIZE]
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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[SIZE=4][B]Appendix 1: Definition of Terms[/B][/SIZE]
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 in the 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 meter used 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."
[SIZE=4][B]Appendix 2: SMPTE Practice[/B][/SIZE]
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 dB FS (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 2 dB. Using pink noise at an RMS level of -20 dB FS 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 -18 dB 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 dB FS. 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 analyser can be used to make the critical 1/3 octave bands equal and reading ~68 dB SPL, yet totalling the specified 83 dB SPL.
[SIZE=4][B]Appendix 3: Detailed Specifications of the K-System Meters[/B][/SIZE]
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.
Other loudness-determining methods are optional. The suggested average section of Meter K-20/LEQA has a non-flat (A-weighted) frequency response, and response time with an equal-weighted time average of 3 seconds. The average section of Meter K-20/Zwicker corresponds with Zwicker's recommendations 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.
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.
Averaging section:
An additional pointer above the moving average level represents the highest average level in the last ten seconds. An "average hold/release button on the meter changes this pointer to an infinite "highest average" hold until released. The RMS calculation should average at least 1024 samples to avoid an oscillating RMS readout with low frequency sine waves, but keep a reasonable latency time. If it is desired to measure extreme low frequency tones with this meter, the RMS calculation can optionally be increased to include more samples, but at the expense of latency. After RMS calculation, the meter "ballistics" are calculated, with a specified integration time of 600 ms to reach 99% of final reading (this is half as fast as a VU meter). The fall time is identical to the integration time. Rise and fall times should be exponential (log).
The various psychoacoustic versions of the K-System meter (e.g. LEQ-A and Zwicker) will be further defined by the implementation. However, the 0 point on all the meters must continue to correspond with 83 dB SPL so that the loudness of the pink noise calibration signal will be the same across all versions of the meter.
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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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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 0 dB on the meter yields 83 dB SPL (per channel, C-Weighted, slow speed). 0 dB 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 0 dB (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 dB FS at 0 VU, and mix normally with the analog console and VUs. However, mixing consoles should be retrofitted 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 multitrack 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 computer. 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 the PC-compatible. 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, 3/11/01, 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 20-bit systems have 24 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 when 24-bit media reach the consumer. Until then chaos will remain in the classical field, and perhaps only metadata will sort out the classical music situation at the listener's end.
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.
[SIZE=4][B]VIII. An Extendable System[/B][/SIZE]
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.
[SIZE=4][B]IX. Metadata and the K-System[/B][/SIZE]
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-Systemfor 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.
[SIZE=4][B]X. In Conclusion[/B][/SIZE]
Let's bring audio into the 21st century. The K-system is the first integrated approach to monitoring, levelling practices, metering and metadata.
[SIZE=4][B]B: Multichannel[/B][/SIZE]
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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[SIZE=4][B]Appendix 1: Definition of Terms[/B][/SIZE]
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 in the 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 meter used 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."
[SIZE=4][B]Appendix 2: SMPTE Practice[/B][/SIZE]
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 dB FS (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 2 dB. Using pink noise at an RMS level of -20 dB FS 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 -18 dB 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 dB FS. 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 analyser can be used to make the critical 1/3 octave bands equal and reading ~68 dB SPL, yet totalling the specified 83 dB SPL.
[SIZE=4][B]Appendix 3: Detailed Specifications of the K-System Meters[/B][/SIZE]
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.
Other loudness-determining methods are optional. The suggested average section of Meter K-20/LEQA has a non-flat (A-weighted) frequency response, and response time with an equal-weighted time average of 3 seconds. The average section of Meter K-20/Zwicker corresponds with Zwicker's recommendations 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.
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.
Averaging section:
An additional pointer above the moving average level represents the highest average level in the last ten seconds. An "average hold/release button on the meter changes this pointer to an infinite "highest average" hold until released. The RMS calculation should average at least 1024 samples to avoid an oscillating RMS readout with low frequency sine waves, but keep a reasonable latency time. If it is desired to measure extreme low frequency tones with this meter, the RMS calculation can optionally be increased to include more samples, but at the expense of latency. After RMS calculation, the meter "ballistics" are calculated, with a specified integration time of 600 ms to reach 99% of final reading (this is half as fast as a VU meter). The fall time is identical to the integration time. Rise and fall times should be exponential (log).
The various psychoacoustic versions of the K-System meter (e.g. LEQ-A and Zwicker) will be further defined by the implementation. However, the 0 point on all the meters must continue to correspond with 83 dB SPL so that the loudness of the pink noise calibration signal will be the same across all versions of the meter.
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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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