Mastering Articles/Texts/Tutorials/Resources

Discussion in 'Mixing and Mastering' started by Mundano, Feb 26, 2018.

  1. Mundano

    Mundano Audiosexual

    Dec 1, 2015
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    i am about to master my project and have found some good articles. I think they can benefit you too. Please post here your links to articles/texts/tutorials/resources and KEEP IT CLEAN, so that others can benefit from searching this thread.

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  3. Mundano

    Mundano Audiosexual

    Dec 1, 2015
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    Loudness Processing Best Practices, Chapter 1 : Loudness Measurement (part 1)
    By Jie Yang (Digimonk), Jun 6, 2017 11:19:05 AM

    Translator's Note:

    This series highlights audio best practices from China. China topped the world gaming revenue charts in 2016. And the Wwise Tour 2016 China stop not only featured popular (50 million daily active users) Chinese games using Wwise, but also boasted over 200 attendees from the local game audio community. Therefore, we were certainly intrigued to take a deeper look into game audio practices in China. By translating blog articles by one of the most influential audio designers within the Chinese gaming industry, we aim to help better understand the audio community and the culture of this vast territory. To the best of our knowledge, this would be the first-ever effort in translating Chinese audio tech blogs into English.

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  4. Mundano

    Mundano Audiosexual

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    Loudness Processing Best Practices, Chapter 1 : Loudness Measurement (PART 2)
    By Jie Yang (Digimonk), Jun 13, 2017 1:37:21 PM

    In his previous blog, Loudness Processing Best Practices, Chapter 1: Loudness Measurement (part 1), Jie Yang (Digimonk) explored challenges, considerations, and compromises surrounding audio standards for various platforms and content types. He also covered a few examples and comparative readings of different measurement units and tools. In this blog, Jie Yang (Digimonk) covers how to make sense of the various options and standards, to be able to make better decisions for our game audio projects and effectively create better and more enjoyable user experiences.

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  5. Mundano

    Mundano Audiosexual

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    Loudness Processing Best Practices, Chapter 2 : Loudness, dynamics and how to process them.
    By Jie Yang (Digimonk), Sep 26, 2017 3:23:52 PM

    Translator's Note
    This is the second chapter of the Loudness Processing Best Practice for Games article series by the Chinese game audio designer Digimonk, originally published on .China topped the world gaming revenue charts in 2016. And the Wwise Tour 2016 China stop not only featured popular (50 million daily active users) Chinese games using Wwise, but also boasted over 200 attendees from the local game audio community. Therefore, we were certainly intrigued to take a deeper look into game audio practices in China. By translating blog articles by one of the most influential audio designers within the Chinese gaming industry, we aim to help better understand the audio community and the culture of this vast territory. To the best of our knowledge, this series has been the first-ever effort in translating Chinese audio tech blogs into English.

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  6. Mundano

    Mundano Audiosexual

    Dec 1, 2015
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    Loudness Processing Best Practice, Chapter 3: Scalable Loudness Processing for Games
    By Jie Yang (Digimonk), Nov 7, 2017 12:26:01 PM

    Translator's Note:
    This is the finale of the three-piece series Loudness Processing Best Practice for Games by the Chinese game audio designer Digimonk, originally published on Closing the series with a post-mortem case study of a released Chinese project, we hope that our effort has helped readers gain a first look at game audio practice in China, as a massive yet under-explored market. Please stay tuned for more in the future.

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  7. Mundano

    Mundano Audiosexual

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    10 Tips for Mastering if You're Not a Mastering Engineer
    by Nick Messitte, iZotope Contributor

    February 7, 2018


    Maybe you’re singer songwriter who has mixed a song and wants to release it quickly. Or perhaps you’re an EDM producer working at a fast clip, and you don’t want to shell out the money to bring every new track to a competitive level. Maybe still you’re a mixing engineer looking to give a client a pseudo-master—not just any pseudo master, but one that sounds better than the average limiter-slam.

    That last guy used to be me. And it was frustrating, because I had tools at my disposal that I knew mastering engineers used regularly. I watched tutorials; I talked to my friends in the mastering world; I read books on the subject.

    I suspect many of us have.

    What follows are the tricks and tips I’ve gleaned over the years to master one track at a time. The heavier work of mastering—blending together a cohesive album; creating DDPs and whatnot—that’s for other articles.

    If you’re a musician, songwriter, producer, or mix engineer looking for tips on how to deliver your own master, read on; you might very well find something useful in my process.

    1. Consider your destination
    You wouldn’t build a house without checking out the neighborhood first, would you? Otherwise you might end up with something incongruous, like a yurt in the middle of some McMansions, or the Guggenheim Museum in the middle of the Upper East Side of Manhattan.

    Similarly, you’ll want to have an idea of where your song will end up before you master it—especially these days, as Spotify, YouTube, Tidal, and Apple Music have embraced (to various degrees) loudness normalization. This means that if you deliver a master at a higher level than a streaming service’s target, the service will lower your song’s level to match the rest of their tunes.

    Chances are, if you’re mastering one tune for consumption, you’re going to stream it—so use this to your advantage: Streaming services tend to operate between -12 and -16 LU, depending on the service, and this gives you more room to create a breathing, musical master.

    You can see how loud your master is by checking your meters often in a plug-in such as iZotope Insight, observing your short term and integrated loudness readings to see if you’re in the ballpark.

    Two outliers here are SoundCloud and Beatport, which both boast no loudness standards and offer lower-quality audio than the other services. Here you might want to master more aggressively, as these services do not limit the loudness of recordings you submit.

    There is also the context of your genre to keep in mind, which brings us to our next point.

    2. Secure genre-appropriate references
    Your next move is to locate tunes similar in tempo, genre, and arrangement density. Hopefully your own music collection serves you well here, or else you might be nickel and dimed to death on downloads. If you mixed the tune previously, you probably have the benefit of a few references already.

    The reason you want a reference track is because your master needs to compete on a commercial platform. It doesn’t need to match a reference perfectly, but the two must meet for a game in the same ballpark. Their overall level, frequency content, and dynamic heft must gel. Think, “would this work on the same playlist?”

    Try to obtain a lossless file if you can, such as .WAV. Lower res file types like .AAC or .MP3s will give you a picture of how a song should sound, but once you’re getting granular, you’ll notice the draining quality of lossy codecs, and the comparison will be harder to achieve as a result. Try to avoid them unless it’s the only option.

    3. Get yourself a meter
    Some people eschew meters, but since you’re mastering out of sheer necessity, you don’t have that luxury. You’ll need an objective check on your choices.

    LUFS Meters: Displaying information in LUs (loudness units), these meters can measure the short term (momentary) as well as integrated (average) level of your track. Integrated loudness is typically used to ensure compliance with broadcast standards, but is useful for spot checking the overall loudness level of your music tracks as well. Short term loudness is useful for checking the dynamic range between the loudest and quietest sections of your mix.

    Spectrum Analyzers: These display the frequency content of your mix, and can be useful in discerning where you need to add low end, take away from the upper mids, and more. They are a good objective check on your ears, and can clue you into what your room might be obfuscating. For more of an explanation, check out the metering section of this article.

    Phase Correlators: This tool gives you a good feeling for whether your master is as wide as your reference, or conversely, if it’s entering dangerous territory, which can happen if you’re employing widening trickery.

    I also tend to check my master in mono frequently, listening to whether important elements have disappeared. If they have, that’s a clue something is awry.

    Loudness History Graphs: if you have a meter that gives you a pictorial view of your loudness over time (such as the one pictured below), this can give you an excellent readout of your dynamic range.

    Luckily iZotope makes a metering system that handles all of these—and in a customizable interface to boot. I speak, of course, of Insight, which I use on every mix and master. It’s a hell of a plug-in.


    Insight Loudness History with real-time labels and indicators

    4. Limit yourself (at first) to 3 tools

    For your purposes, start with these three tools: EQ, compression, and peak limiting at the end of the chain. That’s it. That’s all you likely need. Whether you put the EQ before the compression or after is up to you—I tend to go EQ before, but that’s me.

    Limit yourself to these, and you’re in far less danger of messing things up. Seriously, there’s a reason Ozone’s Master Assistant doesn’t automatically slap stereo-width tools and spectral shaping on your master, and it’s because they are garnishes, not staples.

    Much fancy trickery can actually be accomplished with these three tools. Take stereo width adjustments: Any EQ with mid/side capabilities can get help you widen or narrow your mix. Simply raise the frequencies you’d like to hear on the sides with broad, wide Q’s to achieve a boost in width. Be careful, of course, because you can easily change the mix’s intention. But the difference between a multiband stereo-width plug-in and an EQ used well in M/S mode is not as big as you’d think.

    4. Try to work in reverse
    Working in reverse means you try to achieve the desired level first, then apply dynamics-shaping as needed, and move on to EQ after that.

    Why work this way? After all, many tutorials promote the opposite, advocating EQ moves first, followed by compression, and then bringing everything up with the limiter.

    I used to work that way. I also used to deliver masters that sounded far more squashed than the ones I deliver now. Also, the choices I made regarding frequency-shaping and dynamic movement didn’t hold up when the overall level was boosted at the end—I found the moves I made at quieter levels sounded off, or the limiter’s innate aesthetic drastically impacted the timbre, undoing all my precious work.

    When I reversed the order of operations, everything changed. The process became quicker, and the results tended to sound more natural, so much so that people began to hire me for the occasional mastering job.

    Yes, you do lose that dramatic moment of making the tune louder at the very end of the process, but c’mon, that was always a cheap thrill anyway. You also get a sense of a song’s innate loudness potential earlier in the process—and believe me, every tune has one. This impacts your choices down the line, giving you more room to operate and more of a structure to work within.

    But if you’re limiting straight away, won’t you hit the limiter too hard? Won’t you introduce distortion into the signal immediately? Sure, you may notice some distortion at this phase. But now you can use the other tools at your disposal—the compressor and the equalizer—to avoid these distorted artifacts. This, in conjunction with corrective tonal choices, becomes the crux of your work.

    Onto compression, which you’ll use in one of two ways: to control the dynamics of the material, or to add color. For tips on how to achieve the former, see this article—particularly the section on how to fine tune attack, release, ratio, and threshold controls. Both processes should help you tame distortion before the limiter, as the dynamic range will be addressed either way, and the limiter won’t have to work as hard.

    Now onto EQ: As you presumably have some knowledge of how to mix, your mix probably sounds good already. Your job now is not to make it sound better, but to allow the mix to shine across all sorts of listening environments and platforms. You do this, as a rule, by implementing very subtle moves—a 1 dB cut at 800 Hz or so if its nasal, a subtle shelf of 1 dB at 8 kHz if it’s too dull, and so on.

    As you can use a compressor to keep the peak limiter from doing heavy lifting, you can also use the equalizer to back off frequency ranges that cause the limiter to clamp down in a particularly distorted manner. You can better do this by implementing the following suggestion.

    5. When monitoring, level match your master to your mix

    A plug-in suite like Ozone 8 will allow you to do this handily, but if you’re mixing and matching plug-ins, try the following:

    Put an unprocessed copy of your mix on a dedicated track. Feed the track and your master-in-progress to the same auxiliary channel, and slap a loudness meter on this aux. Next, get your master in the loudness-ballpark.

    Now, as you loop the loudest section of the song, note the loudness reading of the unprocessed mix, switch to the mastered track, and bring its fader down till you’ve achieved the same meter reading as the mix.

    (Note that I said level match your master to the mix and not the other way around—if you were to boost the mix, you’d hit the digital ceiling and hear nasty distortions.)

    As you listen to the two tracks at comparable levels, see if what you’ve done has been an improvement, or if it’s made things worse. This might not be the best moment to judge EQ aesthetically, but you’ll surely hear the differences in dynamic range: if something sounds too squashed, it’ll jump out here, and you can back off your compressor, or make limiter-based equalization decisions (i.e, backing off 100 Hz or so if the kick triggers the limiter too hard).

    As soon as you like your results, move on to the next step, which is:

    6. Level match your reference track to your master

    Much like you did before, import a reference track into Ozone 8, or set up another track with a reference mix on it. Now, level match that reference track to your master.

    As you switch between the reference track and the master, you can make aesthetic decisions regarding EQ. Does the reference feel brighter than your master? Then go ahead and try to match it with a subtle shelf. Does it have cleaner low mids? Try a dip in midrange to uncloud the master. Check compression as well: Does the master feel more squashed than the reference mix? Pull back on the compressor, or don’t hit the limiter so hard.

    7. Translation is key

    It’s a mixing engineer’s job to make the mix good as it can be. A mastering engineer’s job is different in one key way: the mix (or group of mixes) must sound good in as many disparate rooms as possible. Thus, you must make sure your master translates to various places.

    How, as a novice, can you do this? By employing every monitoring environments at your disposal. Work off your trusted rig, for sure, but then monitor the mix any which way you can, including through studio cans, computer speakers, laptop speakers, clock radios, your car, your best pair of consumer headphones, and your worst pair of earbuds. Buy terrible speakers for the sole purpose of monitoring in terrible conditions. Seriously—it’ll go a long way.

    Take the average here and look for acceptability: if you’re consistently finding everything harsh on consumer grade headphones, well, you need to tame that high end. If your midrange is underrepresented on all your smaller speakers, you need to give it some juice. And so on.

    8. Monitor consistently
    For your main setup, it’s quite important to have a fixed monitoring level, as listening at different levels can lead to inconsistent choices. You might boost the lows when listening quietly, then turn up the level, decide the mix is too boomy, and wind up boosting the highs; you could’ve left everything alone in the first place!

    Your monitoring level should be comfortably loud—as in, loud enough to hear the effects of the lows and trebles, but not so loud that you fatigue your ears. Some engineers prefer 80–83 dB SPL. Whatever you choose, a fixed monitoring level will help you make objective decisions, because it will be an immutable, stable point of reference in your studio. You’ll begin to know, over time, that if a master sounds good at this level, it will translate to other systems at other levels.

    It’s also useful to have monitoring level about 12 dB quieter on hand (often called a dim position), to make decisions about how the master sounds when listened to more quietly. Flick to this setting occasionally, and then come back. You can set one up in your DAW with any simple gain plug-in; just be sure to leave it off when you bounce.

    9. Rinse and repeat
    Mastering engineers, as a rule, work very fast. It makes sense: working quickly helps you stay objective—you don’t find yourself diving into the weeds if you actually avoid the weeds. One way to work fast and achieve good results is to cycle through these steps again and again, making quick changes in short order.

    It could look like this: I’m happy with the way the mix is slamming, but the master feels a little thick in the low mids. So I make a change to the EQ. Now, I check it against the reference track, and I find they’re in the same ballpark. I proceed to check the master against my unprocessed mix, but notice my low-mid dip has had an adverse effect on the compressor. Here, maybe I adjust the comp’s threshold, or I change its sidechain filter a bit so that different frequencies are triggering compression.

    It goes on and on until I walk away, come back ten minutes later, and find myself satisfied with the results.

    10. Export and Dither
    Once you’ve gotten the master how you like it, it’s time to bounce, render, or export the file. There are a few considerations here, and once again they depend on your destination. For CD and the majority of online aggregators, the name of the game is 44.1 kHz / 16-Bit WAV files. For all else, I recommend brushing up on this article.

    You’ll want to apply dither to the final master, and if we had another 3,000 words, I could explain why. We don’t so I’ll just refer you here. For what it’s worth, I tend to export my files in their original sample rate and resolution, and then apply sample rate conversion, add dither, or export to lossy codecs from a standalone application like RX. This gives me freedom to move in any direction I choose, as well as excellent SRC tools.

    You may notice that we didn’t touch much upon stereo-width processing, multiband compression, expansion, or other tools newly available, like Ozone’s Master assistant, which makes for another great reference point.

    Why did we save these topics for other articles? For one thing, we don’t have the space, and for another, you most likely can get a great-sounding master without these tools. I, for one, believe that you can—after all, you’re an iZotope reader, so you’re passionate about making informed decisions all along the chain, from the mixing to mastering process. It’s our hope this article will provide another useful perspective on your journey to audio mastery.

    Yes, that’s another terrible pun for you.
    Last edited: Feb 26, 2018
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  8. Mundano

    Mundano Audiosexual

    Dec 1, 2015
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    you can apply the same principles with your own preferred FREE software

    Audio Mastering Tips & Tutorials
    Learn How to Master Audio from iZotope
    iZotope's critically-acclaimed Ozone mastering software has been a studio staple for audio engineers and music producers. Learn about audio mastering and how to create professional-sounding master recordings using the tutorials below.

    Watch Webinars
    Mastering Month Series: These webinars were part of iZotope's Mastering Month. Each topic features insights from noted mastering engineer and iZotope Education Director Jonathan Wyner.

  9. Mundano

    Mundano Audiosexual

    Dec 1, 2015
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    Reaper Render Modes


    Render speed options are full-speed offline, 1x offline, and online.

    Online means that the audio system is on, and playback happens exactly as if you were not rendering. You always need to use online render if you have hardware returns, ReaInsert, or anything else that is not completely "in the box."

    Offline means that the audio system is off, so there is no overhead caused by having to sync with the interface driver. In theory this means that every element of the render can go exactly as slow or as fast as it needs to, in order to get its samples rendered.

    1x offline is just like offline, except that the render is not allowed to go faster than realtime. It can still go slower than realtime, if needed.

    Keep in mind that when you are rendering, you are writing to disk, so you are using more resources than just playing back the project. If your project is almost disk-throttled due to playing back lots of media or using disk-streaming sampler plugins, it's possible for an online render to fail, because of the additional disk writing caused by the render. This should not happen when using an offline render, because the render can go slower than realtime, in order to wait for the disk to be available.

    Also, as mentioned by earlier posters, some plugins depend on playback happening at exactly 1x speed (essentially, plugins that look at the clock to see what time it is, rather than counting the number of samples that are passing through). This can cause full-speed offline renders to sound different from real-time playback.

    So unfortunately there's no answer to what is the safest render method -- it depends on the project. If your project plays back comfortably within the limits of the available CPU and disk I/O, all render methods should give an exact reproduction of what you are hearing. If you have hardware returns, you need to use online render. If you have insufficient system resources for your project to play back while simultaneously writing the rendered audio, offline render is safer than online, because the render can take as much time as it needs to wait for the disk. If you are using plugins that don't support faster-than-realtime playback, you need to use either online or 1x offline, for the render to sound exactly like playback. If you are both disk-throttled and using plugins that require realtime playback, you may need to render or freeze certain tracks before rendering the whole project.
  10. Mundano

    Mundano Audiosexual

    Dec 1, 2015
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    Level Practices (Part 1)
    Part I: The 20th Century
    Dealing With Peaks

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

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

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

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

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


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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

    • Am I interested in maintaining headroom and avoiding peak clipping or do I want the highest possible signal-to-noise ratio at all times?
    • Do I need to simplify dubbing practices or am I willing to require constant supervision during dubbing (operator checks levels before each dub, finds the peaks, and so on)?
    • Am I adjusting levels or processing dynamics–mastering for loudness and consistency with only secondary regard for the peak level?
    Consider your typical musical sources. Are your sources totally digital (DDD)? Did they pass through extreme processing (compression) or through analog tape stages? Pure, unprocessed digital sources, particularly individual tracks on a multitrack, will have peak levels 18 to 20 dB above 0 VU. Whereas processed mixdowns will have peak-to-average ratios of up to 18 dB (rarely up to 20). Analog tapes will have peak levels up to 14 dB, almost never greater. And that’s how the three most common choices of translation numbers (-18,-20, and -14) were derived.

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

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

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

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

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

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

  11. Mundano

    Mundano Audiosexual

    Dec 1, 2015
    Likes Received:

    Level Practices (Part 2)
    Part II: How To Make Better Recordings in the 21st Century – An Integrated Approach to Metering, Monitoring, and Leveling Practices.

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

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

    A: Two-Channel

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

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

    Summary of VU Inconsistencies and Errors

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

    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.

    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. [​IMG]The top 50% of the physical scale is devoted to only the top 6 dB of dynamic range, and the meter’s useable dynamic range is only about 13 dB. Not realizing this fundamental fact, inexperienced and experienced operators alike tend to push audio levels and/or compress them to stay within this visible range. With uncompressed material, the needle fluctuates far greater than the perceived loudness change and it is difficult to distinguish compressed from uncompressed material by the meter. Soft material may hardly move the meter, but be well within the acceptable limits for the medium and the intended listening environment.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

    The three K-System meter scales are named K-20, K-14, and K-12. I’ve also nicknamed them the papa, mama, and baby meters. The K-20 meter is intended for wide dynamic range material, e.g., large theatre mixes, “daring home theatre” mixes, audiophile music, classical (symphonic) music, “audiophile” pop music mixed in 5.1 surround, and so on. The K-14 meter is for the vast majority of moderately-compressed high-fidelity productions intended for home listening (e.g. some home theatre, pop, folk, and rock music). And the K-12 meter is for productions to be dedicated for broadcast.
    Note that full scale digital is always at the top of each K-System meter. The 83 dB SPL point slides relative to the maximum peak level. Using the term K-(N) defines simultaneously the meter’s 0 dB point and the monitoring gain.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

    Here is the K-20/RMS meter in close detail, with the calibration points.
    Individuals who decide to use a different monitor gain should log it on the tape (file) box, and try to use this point consistently. Even with slight deviations from the recommended K(N) practice, the music world will be far more consistent than the current chaos. Everyone should know the monitor gain they like to use.

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

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

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

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

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

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

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

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

    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.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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


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

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

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

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

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

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


    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.

  12. Mundano

    Mundano Audiosexual

    Dec 1, 2015
    Likes Received:

    Mastering Techniques
    What is Mastering?
    “Mastering” is the process of preparing a sound track for distribution. It might involve, but is not limited to:

    • Critical listening, to verify that there are no technical errors in the final product.
    • Fixing equalization problems that were caused by poor acoustics in the original production/mixing room.
    • Applying final tone control to match the overall tone of other tracks in the collection, or comparable tracks in the same genre.
    • Applying final loudness adjustments to match the level of other tracks in the collection, or comparable tracks in the same genre.
    • Sequencing of tracks into a collection ( i.e. sequencing songs on a CD ).
    • Applying frequency-limiting and dynamic-range adjustments, to match the media that will be distributed.
    • Creation of metadata (such as track markers, song name(s), performance rights, etc ) for the distribution media.
    • Exporting/packaging the distributable in a format that is suitable for the next stage of duplication/distribution.
  13. mercurysoto

    mercurysoto Audiosexual

    Nov 23, 2011
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    The bottom of the rabbit hole, next to Alice's
    This article has made my day. Really insightful. Thank you a lot. Begin with the limiter and then shape the balance for it... genius.
  14. Mundano

    Mundano Audiosexual

    Dec 1, 2015
    Likes Received:

    Loudness Explained
    What is Loudness and Why is it important?

    Today, the most fundamental audio issue of all is control of loudness. Every day, millions of people adjust their volume controls over and over. Music recordings from the past often appear to be significantly softer than modern Pop and Rock recordings, and in a television context, promos and commercial are generally much louder than e.g. film, drama or newscasts. No wonder that it is always the volume buttons on remote controls that get worn out the first!

    Since the early days of digital audio, the most common way of determining the level of a given piece of audio has been to measure sample-peak level. However, this method is easily deceived and in the effort to appear louder than competitors, many producers and mastering engineers have found it necessary to use excessive amounts of compression, limiting and maximization, which not only make audio highly inconsistent in terms of loudness (compared to e.g. older recordings and across genres), it also compromises the quality of the program material significantly.

    There is a solution
    Rather than counting the samples, level should be measured by how loud the listener perceives a given piece of audio - in other word s,Perceived Loudness in combination with a new, improved way of measuring peaks called True-peak is the solution to the problem. For this purpose, a number of international broadcast standards have been developed based on thorough research and circumstantial listening tests performed by independent organizations such as Communications Research Centre (CRC) and McGill University in Canada. Further, expertise from external research institutes and manufacturers in the film and music industry - including Dolby and TC Electronic - has been brought into the equation as well. For an overview and in-depth explanation of the various broadcast standards, please click here: Broadcast Standards

    Image Legislation
    In fact, e.g. notoriously loud commercial blocks in television have caused legislative assemblies across the globe to make compliance with certain broadcast standards mandatory. For instance, The CALM Act has been passed in the US, which demands US broadcasters to comply with the ATSC A/85 standard, and in Italy EBU R128 compliance has already been turned into law, while broadcasters several other European countries such as France, Germany, Switzerland, Austria, Norway and Spain aim to comply with R128 in production, ingest and transmission.

    Many other countries throughout the world are currently in the process of legislating in this particular field, which is a clear sign of how serious the issue has become in digital Nand with multi-platform broadcast. In other words, there is no doubt that this is the way of the future for broadcasters.

    What it All Comes Down To...
    To sum up, audio is precious and deserves to be reproduced respectfully. For ages, sound was a natural phenomenon, only existing in the exact moment it was being produced, but technology allowing for recording and reproduction of audio has changed that once and for all. Now, beautiful audible moments can be captured and reproduced to enjoy at any time. However, technology can also be abused, which, as described in the above, is rarely beneficial to the music and film-loving listener. For example, excessive and inexpedient use of compression, limiting and maximization causes audio to suffer considerably.

    With the new broadcast standards - and the equipment that allow for compliance with these standards - production, post and broadcast professionals now have a valuable and efficient set of tools in the ongoing fight against the Loudness Wars. With the new broadcast standards, cross-genre program material can finally co-exist, and volume knobs and buttons can expect a longer life, while audiences will get a far more pleasant listening experience. Everybody wins!

    Loudness Essentials
    • K-Weighting
    • LKFS,LUFS & LU
    • Loudness Range, Program Loudness & Descriptors
    • Gating
    • Target Levels
    • True-peak
    K - Weightlifting
    Rather than trying to measure audio level by counting samples (sample-peak level), circumstantial research has proven that even though two pieces of audio may be measured to be equally loud using the sample-peak method, they may very well be perceived as being very different in terms of level. Studies based on substantial listening tests performed by independent organizations such as Communications Research Centre (CRC) and McGill University in Canada have helped developing a method to measure audio level based on perceived loudness. Without getting into the technical details, a so-called K-weighted filter curve (based on the above-mentioned research results) is applied to each audio channel, which basically builds a bridge between subjective impression and objective measurement.

    The K-weighting method is an essential part of a global, open standard defined byThe InternationalTelecommunication Union: ITU 85.1770 (now updated to BS.1770-3).

    When measuring loudness, three terms are essential to be aware of: LKFS, LUFS and LU. What tends to create confusion is that these terms are very similar and basically aims at describing the exact same thing.

    LKFS is an abbreviation of: Loudness K-weighted Full Scale, and one unit of LKFS is equal to one dB.The LKFS term is used in the ITU BS.1770 standard and the ATSC A/85 standard also operates with this term. Other organizations, such as The European Broadcast Union (EBU), uses the term LUFS, which is an abbreviation ofLoudness Units Full Scale. Despite the different names, LF KS and LUFS are identical. Both terms describe the same phenomenon and just like LKFS, one unit of LUFS is equal to one dB.

    LKFS/LUFS are absolute measures, and depending on which broadcast standard is in use, the loudness target level could be e.g. -24 LKFS or -23 LUFS. However, in order to aim for a more 'traditional number, a relative measure has been defined: Loudness Units (LU). Now, the broadcaster can set the target level (regardless of whether it is -23 or -24) to 0 LU, and again, one LU is equal to one dB.

    Loudness Range, Program Loudness & Descriptiors
    Loudness Range - or LRA - describes the overall program material range: From the softest part to the loudest part The range is quantified in LU, and to avoid extreme events from affecting the overall result, the top 5% and the lowest 10% of the total loudness range is being excluded from the LRA measurement. For example, a single gunshot or a long passage of silence in a movie would result in a very broad Loudness Range even though it would not be representative in the big picture.

    The LRA parameter was originally developed by TC as the descriptor named 'Consistency: Later, it was adopted by e.g. EBU R128 standard and is currently under consideration for implementation at ITU as well.

    Program Loudness aims at describing theaverage program material loudness. Sometimes, Program Loudness may also be referred to as Integrated Loudness. Program Loudness is described using LUFS or LIKES.

    Loudness meters featuring 'EBU Mode' will display the two above parameters, which will be projected as two numeric values - ordescriptors - which will represent a valuable overview of the total 'loudness landscape' of the program material being measured.

    When measuring Program Loudness, merely calculating the average level will not always be desirable as certain events - e.g. long passages of silence (or very soft background noise) in a movie - affect the Program Loudness parameter.Therefore, a gating scheme that pauses the measuring when the audio level drops below a threshold of -10 LU relative to an ungated measurement of the same program material has been developed.

    The benefit of applying this gate is that the measurement becomes far more cross-genre friendly, allowing for example movies and classical music to be 'loudness aligned' with e.g. pop music and commercials. Note that to be efficient in a broadcast context, various types of program material with very different durations - such as a 2 hour movie and a 20 seconds commercial - must be able to be aligned in terms of loudness, and the gate is a very powerful tool in making this possible.

    Target Levels
    Target levels are specified in various broadcast standards, but only vary slightly. For instance, the ATSC AMS standard recommends a target of-24 and uses the LKFS term, whereas the EBU R128 standard sets the target level at -23 and uses the LUFS term. One of the reasons for this difference is that the R128 standard employs the above-mentioned gate, which in effect makes most measurements equivalent to -24 LKFS/LUFS without the gate-yet more useful for aligning loudness across genres.

    Since loudness measuring is based on an algorithm that builds on a study of subjective perception, in theory, program material that complies with the determined LRA and Program Loudness of a certain broadcast standard can in fact overload if normalized the traditional way (quasi-peak or sample-pea k). Therefore, normalization is also part of many broadcast standards, and to comply, broadcasters must use a true-peak meter.

    Many loudness meters have a built-in true-peak meter, and what sets the true-peak meter apart from sample-peak meters is a special algorithm - donated by TC - that not only looks at the actual samples, but also inter sample peaks. In effect, the true-peak mater can unveil peaks in between actual samples that would otherwise cause distortion. Therefore, a true-peak meter actually 'goes beyond 0 dB'. A reading using a traditional sample-peak meter that displays a max of e.g. -0.2 dB could go as far as +3 dB on a true-peak meter reading.

    Please note that this does not indicate acceptance of exceeding 0dB on a true-peak meter, but it provides a more precise reading that helps in normalizing program material without compromising the quality of the audio. As an example, the max value of normalized program material according to the EBU R128 standard is -1 dBTP (d B True-Peak).
  15. Mundano

    Mundano Audiosexual

    Dec 1, 2015
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    Broadcast Standards

    There are numerous broadcast standards. Here is a complete overview, including details of each standard.

    Many countries have already legislated in regard to compliance with specific broadcast standards, while others are still in the legislative process.

    ITU BS.1770-3
    In many ways, The International Telecommunication Union's BS.I 770 recommendation is global and one of the most important broadcast standards as many other standards are based on it.The ETU standard concerns Broadcast loudness and True-peak Level measurement, and the loudness part is based on an Leq measurement employing K-weighting, which is a specific frequency weighting developed by the Communications Research Centre (a federal research institute in Ottawa, Canada).This baseline method is relatively simple, but it is based on excessive listening tests and has been verified independently. The True-peak part of the standard was specified by AES SC-02-01.

    As mentioned, many other broadcast standards are based on ITU BS.1770, Including ATSC A/85 (the US), EBU R128 (Europe), OP-59 (Australia) and TR-B32 (Japan).

    In 2011, the recommendation was released in a revised version: ITU 65.1770-2 that employs a relative gate with regard to Program Loudness measurement, which was adopted from the R128 standard defined byThe European Broadcast Union (EBU).

    In 2012, the recommendation was updated once again to become version BS.1770-3.

    ITU BSA 770 published 2006

    ITU BS.1770-2 published March 2011

    ITU BS.1770-3 published August 2012

    EBU R128
    The P/LOUD group, which is part of The European Broadcasting Union (EBU), has defined the RI 28 standard based on ITU BS.1770. However, the group also added new tools such as a relative gate that ensures even more consistent loudness across genres and types of program material. Some of these tools have been implemented in the updated version of ITU's recommendation: ITU BS.1770-3.

    Basically, the R128 standard builds on 4 tech documents: EBU Tech 3341, EBU Tech 3342, EBU Tech 3343 and EBU Tech 3344.

    EBU Tech 3341
    This document specifies loudness metering within the 8128 domain. Quite simply, a loudness meter is aDowed to use the 'EBU Mode' term if certain criteria are met First of all, three time scales must be available: Momentary (M), Short-term (5) and Integrated (0 - also referred to as Program Loudness My real-time live meter with EBU Mode must be able to display the three time scales, though not necessarily at the same time, and it must also be able to display the maximum value of the Momentary Loudness (reset when Program Loudness is being reset).

    Momentary Loudness must be measured using a sliding time window of OA seconds, wh Ile Short-term Loudness must be measured using a sliding time window of 3 seconds. Program Loudness must be measured using a specific gating method that excludes measurements of parts dropping below a threshold of -10 LV relative to an ungated measurement of the same program material.

    Further, a meter featuring EBU Mode must also be able to display IRA (loudness Range), which is a measurement of the variation of loudness on a macroscopic scale. This parameter is a supplemeni to the overall loudness measurement (ProgramLoudness).

    The terms used for expressing the measurements are LU (Loudness Units) and LUFS, which is the same as LKIS (Loudness K-weighted Full Scale) used by other broadcast standards. The target loudness of EBU's 8128 standard is 23 WI-S. For in-depth specifications, please consult the original EBU Tech 3341 document.

    EBU Tech 3342
    This document specifies audio normalization based on loudness measurements as described in EBU Tech 3342. The average loudness level, or Program Loudness, should be used in combination with Maximum True-peak Level and Loudness Range (LRA) to correct program material in accordance to the RI28 specifications.

    In essence, LRA is determined by analyzing the loudest and the softest parts of the program material. However, the lower percentile of 1096 is being ignored as is the upper percentile of 95% to avoid extreme events such as a single gunshot or long passages of silence to manipulate the overall result in an undesirably way.

    For in-depth specifications, please consult the original EBU Tech 3342 document.

    EBU Tech 3343
    This document describes guidelines for production and implementation in accordance with EBU R128. Both Program Loudness, LRA and True-peak metering is explained and strategies for implementing a loudness strategy at various stages of production are offered.

    For in-depth specifications, please consult the original EBU Tech 3343document

    EBU Tech 3344
    This document describes how to loudness normalize when distributing program material to various end user platforms, including radio, television and portable devices in various formats such as stereo and 5.1 surround.

    For in-depth specifications, please consult the original EBU Tech 3344document.

    Published August 2010

    ATSC A/85
    ATSC A/85 was specified byThe Advanced Television Systems Committee in 2009 and applies to US broadcast digital television. N85 is rooted in the ITU-R BS.1770 Loudness and True-peak level standard. It specifies anchor based normalization for regular programs, but all-source loudness normalization for commercials and interstitials, both at a default Target loudness of -24 LKFS. Put differently, regular programs are under a liberal, rubber-band rule, while the level of commercials is defined precisely and transparently.

    A/85 includes extensive information about calibrated monitoring environments and may function somewhat like a Dolby manual. Unlike EBU R128, A/85 is only focused on the digital television platform and on the AC3 codec.

    In 2011, to give the CALM Act a chance of becoming effective, two revision were published, calling for the inclusion of all sources when measuring commercials, i.e. measuring loudness rather than speech. In the original version of A/85, only the 'anchor method was recognized. Even though the 2011 revisions were published after ITU had defined BS.1770-2, the ATSC standards ambiguously pointed to BS.I 770-1, which was then no longer in effect.

    From March 2013, this ambiguity has gone, because the A/85 recommended practice now prescribes ITU BS.1770-3 for all programming. Hopefully, the proprietary speech detecting measurement algorithm ATSC has come to rely on will be updated accordingly before long.

    11:2-832 is a Japanese broadcast standard that builds on ITU BSI 770-2, which means that a relative gateis employed. However, the target level is -24 LUFS/LKFS as opposed to the -23 LUFS target level of the EBU R128 standard which also employs the gate. As a rule of thumb, a gated measurement of -23 LUFS/LKFS equals an un-gated measurement of -24 LUFS/LKFS.

    Operational practice by FreeTV, Australia. OP-59 is rooted in BS.1770 Loudness and True-peak level and recommends a speech based as well as a universal approach to audio normalization. NI short form programs should be measured using the universal (full mix) method.
  16. Mundano

    Mundano Audiosexual

    Dec 1, 2015
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    Mixing Sound for Film – Audio Post Production, An Overview
    So you’re going to mix sound for film for the first time. You have been an audio engineer for a while. You have spent the years building up your skills, and you love movies! You think to yourself “mixing sound for film seems like a perfect fit”. If those previous statements are true, then it probably is a perfect fit, but there are a few differences between audio post production for film and mixing a record for commercial release.

    The Basics You Should Know Before You Start.
    If you have worked with other people on projects like this before, you may know most or all of the things I will cover in this article, but if you are setting out on your own for the first time, these are the key points which I found incredibly helpful to know.

    Depending on the experience of the director, producer, picture editor, and most definitely the “on location” sound person, your job can vary a bit in its scope. Just know this, mixing music for bands is going to seem a lot easier after you get done mixing sound for film.

    Project Timeline (Scoping)
    You’ve been mixing three to five minute songs for most of your career, and you know how many hours can go into tracking and mixing a fully polished song. How in the world are you going to scope out how long it will take to mix a whole movie? I should warn you here, you are most likely going to be doing a lot more than just mixing. Think dialog clean up, audio restoration, Foley and sound effect design, as well as automated dialog replacement or ADR work.

    Most small budget indie films run around the 30 minute mark, so we will use that as a measure to base your estimate on. I just finished mixing audio and all the other post production elements for a 38 minute film. I put 150 hours into that project, and that was rushing it. There was so much more that I would have loved to have done for the audio mix, but we were on a serious deadline. I think that film could have benefited from another 150 hours of work, but we didn’t have that amount of time.

    I would scope out a post production project in this way:
    For a 30 minute film you will put in between 150 and 300 hours of work depending on the quality of the audio you are given. This particular film I was working on had MAJOR audio problems, so if you are handed pristine quality audio to work with, you may be able to cut that timeline down to perhaps 100-120 hours. Better to give yourself the buffer though. Always under promise and over deliver. If your film is an hour in length, just double the time to 300-600 hours. Now of course this is for pushing-it-through, low-budget films. If your director has the budget, the vision, and the desire, you may end up working for much longer. Keep in mind that Star Wars spent a whole year on audio post production. If you want that level of quality, it will take that amount of time.

    Remind your director and producer in the beginning of the project that they can “have it quick, have it cheap, or have it done right. Pick only two”.

    What Format?
    Be sure to tell your picture editor what file formats you will need. Most likely .mov for picture lock, and an OMF or AAF session export from their editing software. If you can import an OMF or AAF file into your DAW this will save you from having to checkerboard the dialog yourself, and save a ton of time.

    There is a great session format translator that can help to convert many editing session formats into compatible formats for many DAWs called AATranslator. They have a list of what formats they support, and they are very responsive if you have questions. Generally Pro Tools will import OMF and AAF files from Avid Media Composer (AMC) really well, and Logic Pro will take imports from Final Cut, and some from AMC. If your editor works in AMC and you don’t have Pro Tools, you can do either one of two things. One, get them to export from AMC into Pro Tools, and then export an OMF or AAF file from that: Or two, buy a copy of Pro Tools, or just download the demo, and either mix the project in that, or use it as a translator to export the OMF or AAF to the DAW of your choice. Logic Pro will reliably accept an OMF non embedded file from Pro Tools, and will sometimes accept and AAF.

    It is not the end of the world if you can’t take a session export though. Your alternate option is to get the picture lock, and an export of the audio in a stereo file at the rate and bit depth you will be working, and off you go in the DAW of your choice.

    Pro Tools, Logic Pro, Nuendo, are definitely good DAW’s to do post production in. I do most of my post production work in pro tools these days, but I have done a lot of work in Logic Pro, and one movie in Ableton Live. Ableton Live is powerful enough to do the work, and the tools are just fine, but Ableton does not support the OMF and AAF file format, so you will be stuck with manually importing the audio and checkerboarding it yourself. This can be 10 hours of work on a 30-40 minute film.

    Stereo or 5.1 Surround?
    One of the first things to establish is, what does the director or producer expect as a final output? Many low budget indie films are only looking for a stereo mix, but the mid to upper level indies that are really going for the film festivals may want a 5.1 surround mix. Make sure you ask the director and or producer up front what they are looking for and discuss the differences in budget for each. Getting set up for, or renting time in a 5.1 mix room is going to run a little more expensive, and they need to know this. If you already have a 5.1 surround sound mixing room, I would guess you probably already know most of everything I am going to cover in this article, so you can skim through and congratulate yourself on what you have learned as an audio engineer. Perhaps you might even like to share some of your experiences with us in the comments section. For the most part I will focus on what it will take for the first timers to get up to speed on a stereo audio mix for film.

    Sample Rate and Bit Depth
    Find out from your picture editor what sample rate and bit depth they would like the files in for your final export. Generally 48kbps sample rate with a 24 bit depth will be fine, and perhaps the highest that their system can handle. Some can handle a higher sample rate, but not many can handle 32 bit depth. Also find out if they want .aif or .wav or some other session format.

    Get a Good Loudness Meter
    iZotope Insight

    This is more important than you think it is, unless you already know all about broadcast standards, and then it is exactly as important as you suspect. As recording and mixing engineers for albums and singles most of our dealings with mixing and “loudness” is “how loud can you make this track without clipping or squashing it too much”. Every band wants their song to be as loud or louder than everyone else’s. There is much that we could discuss about the “loudness wars” but plenty of other people have already written tons of valuable information on that already.

    Broadcast Standards
    [​IMG]In the case of mixing sound for film or television there are very specific standards that have been established, and passed into law. Cinema, and DVD can be a little more lax than television, but it has been my experience that mixing film audio to the BS 1770-3 a.k.a. A85 standards established by the CALM act works out very well. This kind of tool can guide you during mixing and help you arrive at a balanced smooth sounding mix that won’t blow up anybody’s speakers. You can always mix to the broadcast standards, and then if your client wants it louder make some easy adjustments from there.

    Before we get too deep into this I want to share one little tidbit of info. In case you don’t know it yet, 1db equals 1LU or loudness unit. All the loudness meters measure in LU, so this is handy to know.

    I suggest that you do a little research on loudness meters and pick the one that you think will work best with your style. Waves has WLM Plus, Dolby has one, TC Electronics has one, and iZotope has the Insight Metering Suite.

    I use both the Insight Metering Suite and Waves WLM Plus because they speed up my workflow, give me easy readouts of all the information I want, and have a reputation of being even more accurate than the Dolby meters. I also have had excellent experience with other iZotope and Waves software, so I trust them to deliver a top of the line industry tool.

    The beauty of using a loudness meter like Insight or WLM Plus is that it has all the presets with the standards worked out for you. If you are mixing for the USA it has those, if you are mixing for the UK or Japan, it has those. If your market’s standards are not represented you can make your own preset. I’m in the USA so I chose the simple US meter preset. This gives you a max peak of -2db, a program average target of -24LU and a dynamic range target of 14LU.

    There are a ton of features on the different meters, but that is for another article. Basically, they shows you your targets, and gives you clear readouts of overs and unders, and where you need to go back and refine. A good loudness meter will make your life easy and remove any worry you have of getting your program mix rejected for peaks over the broadcast standards.

    Calibrate Your Control Room
    SPL meter

    What does this mean, and how do I do this? What we mean in this case is to set the actual listening volume in your control room to a standard that will respond to the dynamic range of a mix in a similar fashion to your target venue. In this case a theater environment. There are different calibration settings for different purposes. Here we will describe a calibration that works for the mid to small sized control rooms, and specifically for mixing film and broadcast material.

    How you do this is actually way more simple than it sounds.

    1. Get a full frequency spectrum pink noise .wav file. (Right click to download this pinknoise file).
    2. Drop this file into a free track in your DAW, and set your session to loop this file.
    3. Hopefully you have already gotten a hold of a loudness metering solution like Waves WLM Plus or Izotope Insight. If not, get it now. The reason to use this instead of the regular meters in your DAW is that these dedicated meters are far more accurate.
    4. Adjust the volume of your pink noise track while watching your loudness meter. Set your pink noise to have a max peak of -20db.
    5. Now set up an SPL meter in your listening position with the mic at what would be your ear level. Many SPL meters can be attached to a camera tripod for easy positioning. Set the mode to “slow”, select the readout “high”, and db mode “C”
    6. While the pink noise that you set to -20db is playing, slowly adjust the volume control of your reference monitors. Bring the volume up until your SPL meter is reading 79db. Mark this position on your volume knob or fader. Now slowly bring the volume up to 85db, and mark this position as well. In mid to small rooms do this with both monitors at once. In larger rooms you would do each speaker individually.
    Congratulations you have just calibrated your control room for mixing a film!

    Many in the industry suggest mixing sound for film at 85dB, but in the smaller room environments this can be too loud, causing ear fatigue and other issues. I suggest mixing at 79dB, and when you think you are done, do one or two passes through at 85dB to double check. This should give you a fairly accurate listening environment without the ear fatigue. In larger rooms you may be able to mix at the 85dB level, and in really big rooms you would calibrate each speaker individually. Now when you listen through the program if elements are too loud bring them down, too quiet bring them up. Trust your ears.

    Make Sure That You Get The Final, Final, and I Mean FINAL Picture Lock to Work With!
    Save yourself from nightmares and pulling out your hair by making it clear to the director and editor and anyone else that needs to know, that the file they give you MUST be the final edit of the film. The final picture lock. Do not begin work on the audio until you have assurances from all parties that this is it. As a matter of fact I would make sure to have that in your contract. If you get halfway through the audio work for the film and suddenly the director or editor tells you they are just going to cut a scene or two, you are going to be in a world of hurt trying to go back through and realign all the audio you have worked on with the new picture lock. Needless to say this will add tons of hours to the project, and mean that you will most likely not make any deadlines you set.

    If the editor can supply you with a new OMF or AAF export then the most you will lose is the naming and organization of your tracks, It will still take time to re-import any audio that you have significantly altered. It can be done, but is a pain in the neck and racks up hours really fast.

    Now the Actual Audio Post Production Work & Workflow
    You may have thought that mixing audio for film was just that, mixing. You thought the editor would give you a perfect dialog track, the music director would hand you all the soundtrack songs, and the composer will drop the most perfectly emotional score into your lap. For the major Hollywood films, or a really professional production company you might get that lucky. And for the most part you shouldn’t have to worry about the composer and music director. They should really be the producer’s and director’s problem, and if they chose the right team members, everything will go smooth. The dialog is where you are going to spend a ton of time.

    If your picture editor is experienced and good at audio restoration you may actually get a dialog track with good levels, and already cleaned up. There is a good chance though that you are going to be left with the job of restoring the audio quality of the dialog track. Here is a short list of the tasks you will undertake, and the order they should fall in:

    Importing Session Files
    [​IMG]Most likely your picture editor is going to want to send you an OMF or AAF session file with all the audio data. I have worked in three different DAW’s for post production, Ableton Live, Logic Pro, and Pro Tools, and unfortunately only one of the three imports the OMF and AAF files properly…. Pro Tools. Logic is super close. It imports the tracks and the automation, but not the track names, and that can be super frustrating when you have a large session to work on. Ableton Live doesn’t accept these files at all. Pro Tools shows it’s strength in this area. You import the AAF and there you are with all your tracks laid out with the proper names, and all the automation. This can save you a bunch of time.

    [​IMG]If the price tag on Pro Tools is too steep, Logic would be my second choice for audio post production, but you will have some cleanup and organizing to do before you really get to start working. Another weakness with Logic is that if you have to insert frames because the director or editor decided to make changes to the picture lock, Logic can be a little finicky with the way it handles the frame count and insertion. For better or worse this is another area where Pro Tools excels. Just type in the frame numbers and you are good to go when working with spotting in Pro Tools.

    Checkerboard your dialog
    [​IMG]If you weren’t able to import an OMF or AAF, or you have more than one character’s dialog on one track you will want to separate each character. Cut your dialog up pretty much line by line so that you will have an audio track for each character that can be eq’d, compressed and effected independently.

    Normalize your dialog
    Level out all the dialog so it falls within the target range of -11db to -10db. This is a rough guide point, quiet sections may fall a tiny bit below this to -11.5db, but I suggest not going above -10db for your max dialog peak. Measure this with the dialog solo’d. I tend to group all of the dialog tracks to a bus for control of all of them in case I need it, this can be handy to use as a side chaining source key for tastefully compressing background music.

    Audio restoration
    iZotope RX Advanced

    Once you have normalized all your dialog to that -11db to -10db range, you will most likely find a lot of undesirable background noise. This will vary from generic noise floor hiss, to crew talking in the background, or crew dropping things in the middle of a scene, or crew footsteps in the background during quiet scenes, or large motor vehicles driving by, or planes flying over head, generators or air conditioners kicking on, or even the mic cable not being taped to the boom and rattling against it… I think you get the picture. There is going to be a ton of stuff that you don’t want in your final mix. It is now your job to remove and clean up all this unwanted noise. Some of this you can do with careful gating and EQ, but if you are dealing with this level of clean up I strongly suggest getting a tool specifically made for this job. My favorite is iZotope RX Advanced. You can get the RX regular version, but if you are deep into restoration you are really going to want the advanced version. I wrote a review of it here. This software can make your life much much easier by quickly allowing you to edit out unwanted sound from dialog tracks.

    ADR or Looping
    After all your restoration work you may encounter some dialog that is just irreparable and completely unusable. You will have to get the actors into the studio to re-record their lines. This is called ADR (Automatic Dialog Replacement) or looping. This is a trick in itself as you need the new dialog to sync up with the lip movements on screen. I will write more about the process of ADR at a later date, but in summary, you will need a quiet studio, preferably a larger room to record in, a screen for the actor to watch the scenes on, and preferably a screen with the wave files visible to them as well.

    Foley and sound design
    Now that you have fixed all the problems with your dialog tracks, you can reward yourself by doing some Foley and sound design work. Make those fight scenes really grab your attention, add footsteps, doors closing, car and other vehicle sounds. Have fun in this section. You can record a lot or all of this yourself, or you can search for sound libraries on Google.

    I would suggest that your max peak for your loudest effects be around -5db for the loudest stuff like explosions and gunshots, but mix to your preference.

    And finally, mixing
    Your dialog and sound effects are in place, soundtrack songs have been delivered, and your score has been dropped into the proper timeline, now you polish. I will talk about the levels to hit in the Mix Targets section below.

    That is the basic run down of the task you are about to undertake.

    Audio Mix Level Targets
    One of the most popular questions that I see when people are getting into mixing sound for film is; “What should my levels be?”. One test you can do to answer the question for yourself, or to validate the answer I provide here is to pull a .mov or .mp4 version of a Hollywood large budget movie into your DAW and watch your loudness meters as the movie plays. You can see the levels where the dialog sits at, the music, and the sound effects.

    Here is what I learned from that experience summarized in one easily digestible chunk. All levels dBFS peak.

    • Max peak: -2db (This was absolute on everything that I tested probably because of BS1770-3/A85)
    • Loud sound effects (explosions, gunshots): -3db to -2db
    • Louder soundtrack or score music not competing with dialog max: -5db to -4db
    • Dialog level: -11.5db to -10db
    Use these as reference points. Start with getting your dialog levels first, and then build everything else around that. These are not hard rules (aside from the max peak of -2db) but a foundation on which to build your audio mix. I found this to work consistently well for me, and puts me right in the ball park to be compliant with the broadcast standards established by the CALM act (BS1770-3/A85).

    I wrote this article because I kept seeing people ask the same questions I was asking before I mixed my first film, and I remember the hours and headaches of searching across many blogs and articles trying to find this information. I have attempted to compile into one location all the basics that you need to know to get started, and hopefully have answered many of the questions that were nagging in the back of your mind. I hope I have been helpful.

    If I have left anything out, or got anything wrong, please leave a comment, add to our knowledge, or ask for more detail. Have fun and mix well.

    If you are a film maker and have been reading this article to figure out how to plan for audio post production feel free to contact me with any questions you may have.

    If you are looking for a sound mixer or sound designer, send me the details of your project and I will get back to you with a price quote. Take a look at my Audio Post Production services page for ideas on what questions I will be asking to give you an accurate quote.
  17. Mundano

    Mundano Audiosexual

    Dec 1, 2015
    Likes Received:

    The Secret of the Mastering Engineer
    By Bob Katz

    Mastering is an art and a science. In this acclaimed booklet, Bob Katz shares good advice about monitoring, metering and processing. About listening to the music and supporting it as the road to Nirvana - from one of the true yogis of our industry.

    Published at SOAS 2004, Poznan.


    Table of contents . . . . . . . . . . . . . . . . . . 3
    Welcome . . . . . . . . . . . . . . . . . . . . . . . . 5
    GettingStarted ...................6
    YourRoomYourMonitors ...........7
    Metering ........................8
    DynamicsProcessing ..............9
    Relative Levels, Loudness
and Normalization . . . . . . . . . . . . . . . . 10
    Recipe for Radio Success . . . . . . . . . . 11
    Dither .........................13 Equalization. . . . . . . . . . . . . . . . . . . . . 13
    Sibilance Control . . . . . . . . . . . . . . . . . 15
    Noise reduction . . . . . . . . . . . . . . . . . . 15
    Monitors .......................16
    Advanced Mastering Techniques . . . . . 17
    Reference . . . . . . . . . . . . . . . . . . . . . . 19
    Glossary .......................19
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    We’re committed to delivering music as the artists and sound engineers intend it to be heard. Housed here are the information and tools necessary to create the highest-quality masters for iTunes. Learn more by reading the Mastered for iTunes technology brief.

    Mastered for iTunes

    To hear how music will sound after it’s encoded and to make important creative choices during the mastering process, download these tools. The Mastered for iTunes droplet is a simple, standalone
    drag-and-drop tool that automates the creation of iTunes Plus format masters, allowing for a preview of songs using the same world-class technology used to encode the entire catalog for the iTunes Library.

    Apple Audio Mastering Tools (OS X v10.9 or greater)

    Apple Audio Mastering Tools (OS X v10.8 or earlier)

    Apple’s free digital audio mixing application, AU Lab, can be used as a host application for Audio Unit effects, including AURoundTripAAC Audio Unit, one of the new Mastered for iTunes tools. If you don’t already have Logic or another Audio Unit host application, download AU Lab to get started with auditioning your audio, detecting peaks and clipping, and performing double-blind listening tests.

    AU Lab
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    How to Make Your Mix Sound Bigger

    Published in Pre-Mastering Tips on June 11, 2014

    [​IMG]When mixing your song, you’ll sometimes want to expand the depth and feel of a track or the overall mix. There are certainly ways to do this and so we thought we’d share some common tips and tricks for making your mix sound bigger.

    EQ Up Lows and Highs
    Pull up an equalizer and boost the low end ever so slightly to add a bit of richness to the bass. Likewise, boosting the highs a touch can add more brightness and details. This works on individual tracks or a master track.

    Layer Up
    Adding more layers is one of the easiest ways to bring more texture and depth to your mix. If you work in a lot of samples, you can often copy and paste the MIDI data into a new patch. You can use different layers to bring out different aspects of the sound you’re looking for.

    Like the rich lows on one piano, but prefer the crisp highs on another? Layer them on top of each other and let the best of each shine through. It’s also a great way to make your music sound unique.

    Add Some Reverb
    The right amount of reverb can add space and depth to your mix, especially in the low range. Experiment with different settings to find the sweet spot between depth and an “overproduced” sound.

    808 Kick Drum
    The legendary Roland 808 drum machine has been lauded for its deep kick drum pretty much since it was released in the 80’s. You can download samples of the 808 for free. Try adding an 808 kick layer to your drum track to see if it makes your kick fill out better. We bet it probably will.

    Widen Your Stereo Image
    Sometimes simply making your panning a little more exaggerated can make your stereo image feel a whole lot wider, causing a bigger perceived space for the listener.

    Using these techniques when mixing can add a great deal of added fullness to your mix, which will translate well after the mastering process. The trick is to use them in moderation and to know when it’s best to use each one. Sometimes, less is more.
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    Equivalent Loudness (Leq) and ITU BS.1770-3

    Audio engineers over the last few years have been becoming more concerned with a method of measuring the loudness of audio programme. This interest has been provoked largely by the "Loudness Wars" in which more and more sophisticated dynamic-range compression equipment is being applied to audio material in mastering in order for each new record to sound "louder" than its predecessors.
    Apart from the questionable musical taste of limiting the dynamic range of material in this way, this practice means that music collections have modern material which sounds louder than that recorded 25 years ago, and an evening's music listening (especially if you have catholic taste) becomes an uncomfortable experience as one track "blasts" above the previous one, because it is too loud: or disappears because it is too soft.

    Because one clear application of Stereo Lab is in the hands of the curator of a music collection or archive, the software contains the signal measurement thereby to assist in normalising audio tracks to a comfortable and well engineered level; so that music collections or archives may be compiled with similar loudness. Stereo Lab fully implements Loudness measurement to ITU BS.1770-3 as this is currently the best and most widely accepted method of loudness measurement. See here for our recommendation of target levels.

    What is loudness? Loudness is not the physical magnitude or intensity of a sound: loudness is subjective. It is the perceived magnitude of a sound which is its loudness. Psychologists have determined that the perception of loudness depends on intensity, frequency and duration.
    These three effects (intensity, frequency and integration-time) are incorporated into the design of sound-level meters which are designed to measure something which resembles the human perception of loudness. However, when assessing the effects of discontinuous sound (like music), normal sound measurements like those using a standard sound level meter are pretty useless, because the problem becomes WHEN to measure. Do you measure the peaks, or the quiet moments? The answer is to measure both. In fact, a series of noise measurements is made throughout the duration of the track and an average figure derived.

    Leq - Equivalent Continuous Sound Pressure level
    Audio engineers looking for a method of measuring loudness of audio programme have followed work originally developed for discontinuous noise level measurements known as Leq which is derived mathematically from standard sound level measurements which are taken over a specified period of time at equal intervals.
    Leq is the methodology employed in Stereo Lab to derive an average loudness per track. In fact, the precise figure displayed is Leq(0dBFS); the Leq measurement in decibels relative to 0dBFS (the highest valve in digital audio coding). This figure is displayed in the main interface table along with two further figures: Crest factor for the left channel; and Crest factor for the right channel.

    All audio metrics may be copied to the clipboard and used in other applications (for example a media-management or database application). This is accomplished by right-clicking the audio file in the main interface.


    Crest factor
    The Crest factor of a waveform displayed here is the ratio of peak amplitude to Leq(0dBFS). You will find that these vary greatly depending on the type of music. Symphonic, orchestral music can have a crest-factor of 17dB, whereas hip-hop music has been shown to have a crest factor of only 8dB; see below.

    Crest factor (ratio of RMS level to peaks) for different types of music

    Here are some statistical values for crest factor for different types of music:

    Music Type Crest Factor (dB)

    Symphonic = 17dB
    Opera = 16dB
    Chamber = 14dB
    Jazz = 13dB
    Speech = 13dB
    Folk = 11dB
    Blues = 11dB
    Popular = 11dB
    Heavy Rock = 9dB
    Hip Hop = 8dB
    Sine wave = 3dB

    It's interesting to note a reduction of Crest factor over time, with classical music having a peak to RMS ration some 6dB greater than "pop", and 9dB greater than hip-hop. As stated earlier, this has less to do with musical genre than to do with the insidious use of greater and greater levels of electronic compression which is applied to modern music to ensure a "loud" record and conspicuous radio-play.

    more information here.

    Channel Correlation Coefficient (CCC)
    Stereo Lab uses Principal Component Analysis (PCA) techniques to adjust the parameters applied to the stereo to 5.1 upconversion process. The Channel Correlation Coefficient is derived in this process and is copy-able to the clipboard as explained above.
    A complete list of the audio metrics automatically harvested for each source and destination file is given below.


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