Helpful ReplyRecording Levels

Page: 12 > Showing page 1 of 2
Author
sharpdion23
Max Output Level: -75 dBFS
  • Total Posts : 784
  • Joined: 2009/04/26 18:07:59
  • Location: Vancouver, BC
  • Status: offline
2012/02/28 17:55:49 (permalink)

Recording Levels

What is the recomended recording level? I usually don't go over -6db when recording. Also are there any exceptions?

Win7 pro 64bit*SonarX1 PE 64 bit* AMD Athlon(tm)64 X2 Dual Processor 6000+ 3.00 Ghz* 4GB Ram* 232GB HD* Cakewalk MA-15D* SPS-66 FireWire

Owner of Sonar 6 Studio* Sonar 7 PE * Sonar 8.0 PE * Sonar 8.5.3 PE * Sonar X1 PE *

Link to upload Screens: http://forum.cakewalk.com/tm.aspx?m=1592276


A lot of people are afraid of heights. Not me, I'm afraid of widths.
#1
sharpdion23
Max Output Level: -75 dBFS
  • Total Posts : 784
  • Joined: 2009/04/26 18:07:59
  • Location: Vancouver, BC
  • Status: offline
Re:Recording Levels 2012/02/28 17:58:44 (permalink)
Is there a use of the gain control in sonar when recording?

Win7 pro 64bit*SonarX1 PE 64 bit* AMD Athlon(tm)64 X2 Dual Processor 6000+ 3.00 Ghz* 4GB Ram* 232GB HD* Cakewalk MA-15D* SPS-66 FireWire

Owner of Sonar 6 Studio* Sonar 7 PE * Sonar 8.0 PE * Sonar 8.5.3 PE * Sonar X1 PE *

Link to upload Screens: http://forum.cakewalk.com/tm.aspx?m=1592276


A lot of people are afraid of heights. Not me, I'm afraid of widths.
#2
batsbrew
Max Output Level: 0 dBFS
  • Total Posts : 10037
  • Joined: 2007/06/07 16:02:32
  • Location: SL,UT
  • Status: offline
Re:Recording Levels 2012/02/28 17:59:33 (permalink)
study this

http://www.digido.com/level-practices-part-1.html


then you will be a genius!


Bats Brew music Streaming
Bats Brew albums:
"Trouble"
"Stay"
"The Time is Magic"
--
Sonar 6 PE>Bandlab Cakewalk>Studio One 3.5>RME BFP>i7-7700 3.6GHz>MSI B250M>G.Skill Ripjaws 4 series 16GB>Samsung 960 EVO m.2ssd>W 10 Pro
 
#3
batsbrew
Max Output Level: 0 dBFS
  • Total Posts : 10037
  • Joined: 2007/06/07 16:02:32
  • Location: SL,UT
  • Status: offline
Re:Recording Levels 2012/02/28 18:01:23 (permalink)

Bats Brew music Streaming
Bats Brew albums:
"Trouble"
"Stay"
"The Time is Magic"
--
Sonar 6 PE>Bandlab Cakewalk>Studio One 3.5>RME BFP>i7-7700 3.6GHz>MSI B250M>G.Skill Ripjaws 4 series 16GB>Samsung 960 EVO m.2ssd>W 10 Pro
 
#4
sharpdion23
Max Output Level: -75 dBFS
  • Total Posts : 784
  • Joined: 2009/04/26 18:07:59
  • Location: Vancouver, BC
  • Status: offline
Re:Recording Levels 2012/02/28 18:06:02 (permalink)
Thanks I will take a look at them

Win7 pro 64bit*SonarX1 PE 64 bit* AMD Athlon(tm)64 X2 Dual Processor 6000+ 3.00 Ghz* 4GB Ram* 232GB HD* Cakewalk MA-15D* SPS-66 FireWire

Owner of Sonar 6 Studio* Sonar 7 PE * Sonar 8.0 PE * Sonar 8.5.3 PE * Sonar X1 PE *

Link to upload Screens: http://forum.cakewalk.com/tm.aspx?m=1592276


A lot of people are afraid of heights. Not me, I'm afraid of widths.
#5
Guitarhacker
Max Output Level: 0 dBFS
  • Total Posts : 24398
  • Joined: 2007/12/07 12:51:18
  • Location: NC
  • Status: offline
Re:Recording Levels 2012/02/28 20:59:34 (permalink)
And here I was thinking all I need to do was keep the wave form from crashing into the ceiling......  (just kidding)...  good stuff there in those links.

My goal is to keep some space above the highest peaks but I also don't freak out when a few of them hit the ceiling.....as long as it still sounds good.

My website & music: www.herbhartley.com

MC4/5/6/X1e.c, on a Custom DAW   
Focusrite Firewire Saffire Interface


BMI/NSAI

"Just as the blade chooses the warrior, so too, the song chooses the writer 
#6
droddey
Max Output Level: -24 dBFS
  • Total Posts : 5147
  • Joined: 2007/02/09 03:44:49
  • Location: Mountain View, CA
  • Status: offline
Re:Recording Levels 2012/02/28 21:42:45 (permalink)
I think he's been hacked. I got redirected to some porn site.

Dean Roddey
Chairman/CTO, Charmed Quark Systems
www.charmedquark.com
#7
batsbrew
Max Output Level: 0 dBFS
  • Total Posts : 10037
  • Joined: 2007/06/07 16:02:32
  • Location: SL,UT
  • Status: offline
Re:Recording Levels 2012/02/28 22:28:37 (permalink)
just checked both links, they work fine.

dude, i think you've been hitting the porn sites so often, that they've hacked YOUR system!!
LOL

Bats Brew music Streaming
Bats Brew albums:
"Trouble"
"Stay"
"The Time is Magic"
--
Sonar 6 PE>Bandlab Cakewalk>Studio One 3.5>RME BFP>i7-7700 3.6GHz>MSI B250M>G.Skill Ripjaws 4 series 16GB>Samsung 960 EVO m.2ssd>W 10 Pro
 
#8
droddey
Max Output Level: -24 dBFS
  • Total Posts : 5147
  • Joined: 2007/02/09 03:44:49
  • Location: Mountain View, CA
  • Status: offline
Re:Recording Levels 2012/02/28 22:51:46 (permalink)
They work fine for me now as well. Not sure what happened there.

Dean Roddey
Chairman/CTO, Charmed Quark Systems
www.charmedquark.com
#9
RobertB
Max Output Level: 0 dBFS
  • Total Posts : 11256
  • Joined: 2005/11/19 23:40:50
  • Location: Fort Worth, Texas
  • Status: offline
Re:Recording Levels 2012/02/28 23:02:30 (permalink)
sharpdion23


Is there a use of the gain control in sonar when recording?

No. That only affects playback.
This may be of interest to you:
http://forum.cakewalk.com/tm.aspx?m=1519608
It was written for an older version, but the principles still apply.



My Soundclick Page
SONAR Professional, X3eStudio,W7 64bit, AMD Athlon IIx4 2.8Ghz, 4GB RAM, 64bit, AKAI EIE Pro, Nektar Impact LX61,Alesis DM6,Alesis ControlPad,Yamaha MG10/2,Alesis M1Mk2 monitors,Samson Servo300,assorted guitars,Lava Lamp

Shimozu-Kushiari or Bob
#10
sharpdion23
Max Output Level: -75 dBFS
  • Total Posts : 784
  • Joined: 2009/04/26 18:07:59
  • Location: Vancouver, BC
  • Status: offline
Re:Recording Levels 2012/02/29 00:29:40 (permalink)
Thanks for all the help so far.

@drod. yeah it happened to me too the first time Though it went fine after that. Wonder what that was all about.

@Robert. Thanks I will have a look at that too.

@Bats. Is the record meter on each track from sonar an accurate meter to use? Or is there a better plugin I could get that is accurate.
 
I was reading this from the site you shared:
 
Digital Meters and OVER Indicators
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. 
 
@Guitar. Thanks for the tip!

Thanks!

Win7 pro 64bit*SonarX1 PE 64 bit* AMD Athlon(tm)64 X2 Dual Processor 6000+ 3.00 Ghz* 4GB Ram* 232GB HD* Cakewalk MA-15D* SPS-66 FireWire

Owner of Sonar 6 Studio* Sonar 7 PE * Sonar 8.0 PE * Sonar 8.5.3 PE * Sonar X1 PE *

Link to upload Screens: http://forum.cakewalk.com/tm.aspx?m=1592276


A lot of people are afraid of heights. Not me, I'm afraid of widths.
#11
droddey
Max Output Level: -24 dBFS
  • Total Posts : 5147
  • Joined: 2007/02/09 03:44:49
  • Location: Mountain View, CA
  • Status: offline
Re:Recording Levels 2012/02/29 00:40:53 (permalink)
sharpdion23

Thanks for all the help so far.

@drod. yeah it happened to me too the first time Though it went fine after that. Wonder what that was all about.
Thanks!
Maybe his site is infected then, and their systems' anti-whatevers caught and supressed it. Might be something he'd want to know about.  I checked the source of this page and Cakewalks isn't generating any sort of link referral type stuff that could have gotten hacked here on this site and caused it to generate a bogus link. So the only other thing that you and I have in common is his web site.
post edited by droddey - 2012/02/29 00:43:34

Dean Roddey
Chairman/CTO, Charmed Quark Systems
www.charmedquark.com
#12
droddey
Max Output Level: -24 dBFS
  • Total Posts : 5147
  • Joined: 2007/02/09 03:44:49
  • Location: Mountain View, CA
  • Status: offline
Re:Recording Levels 2012/02/29 00:46:21 (permalink)
I sent him a warning about this.

Dean Roddey
Chairman/CTO, Charmed Quark Systems
www.charmedquark.com
#13
sharpdion23
Max Output Level: -75 dBFS
  • Total Posts : 784
  • Joined: 2009/04/26 18:07:59
  • Location: Vancouver, BC
  • Status: offline
Re:Recording Levels 2012/02/29 00:54:01 (permalink)
@Drod. I've got kasperky 2012......I guess it was a bit slow on that

Win7 pro 64bit*SonarX1 PE 64 bit* AMD Athlon(tm)64 X2 Dual Processor 6000+ 3.00 Ghz* 4GB Ram* 232GB HD* Cakewalk MA-15D* SPS-66 FireWire

Owner of Sonar 6 Studio* Sonar 7 PE * Sonar 8.0 PE * Sonar 8.5.3 PE * Sonar X1 PE *

Link to upload Screens: http://forum.cakewalk.com/tm.aspx?m=1592276


A lot of people are afraid of heights. Not me, I'm afraid of widths.
#14
batsbrew
Max Output Level: 0 dBFS
  • Total Posts : 10037
  • Joined: 2007/06/07 16:02:32
  • Location: SL,UT
  • Status: offline
Re:Recording Levels 2012/02/29 11:40:11 (permalink) ☄ Helpful
so, back on topic:

i personally record almost all of my tracks with no more than -10db peaks.....

with the exception of drums, which i allow to -8db peaks, but i process them at mixdown, and knock the peaks off with compressors and limiters....


i find the closer i get to -8db peak as a MAXIMUM allowable peak....
the more 'crunchy' the sound is.

that actually works for me, with drums, but not with anything else.


if i record something all the way down to -14db peak (which means an average RMS value of about -23), that seems to be the "SWEET SPOT" for my system, signal chain, method of recording, etc.




Bats Brew music Streaming
Bats Brew albums:
"Trouble"
"Stay"
"The Time is Magic"
--
Sonar 6 PE>Bandlab Cakewalk>Studio One 3.5>RME BFP>i7-7700 3.6GHz>MSI B250M>G.Skill Ripjaws 4 series 16GB>Samsung 960 EVO m.2ssd>W 10 Pro
 
#15
sharpdion23
Max Output Level: -75 dBFS
  • Total Posts : 784
  • Joined: 2009/04/26 18:07:59
  • Location: Vancouver, BC
  • Status: offline
Re:Recording Levels 2012/02/29 12:04:30 (permalink)
In the loudness war do they record over? Or is that a mastering only thing?

Win7 pro 64bit*SonarX1 PE 64 bit* AMD Athlon(tm)64 X2 Dual Processor 6000+ 3.00 Ghz* 4GB Ram* 232GB HD* Cakewalk MA-15D* SPS-66 FireWire

Owner of Sonar 6 Studio* Sonar 7 PE * Sonar 8.0 PE * Sonar 8.5.3 PE * Sonar X1 PE *

Link to upload Screens: http://forum.cakewalk.com/tm.aspx?m=1592276


A lot of people are afraid of heights. Not me, I'm afraid of widths.
#16
batsbrew
Max Output Level: 0 dBFS
  • Total Posts : 10037
  • Joined: 2007/06/07 16:02:32
  • Location: SL,UT
  • Status: offline
Re:Recording Levels 2012/02/29 13:16:51 (permalink)
i'm talking individual tracking levels.


loudness war = douchebag.

mastering not equal to mixing.

level control = quality of sound, not volume of sound.


Bats Brew music Streaming
Bats Brew albums:
"Trouble"
"Stay"
"The Time is Magic"
--
Sonar 6 PE>Bandlab Cakewalk>Studio One 3.5>RME BFP>i7-7700 3.6GHz>MSI B250M>G.Skill Ripjaws 4 series 16GB>Samsung 960 EVO m.2ssd>W 10 Pro
 
#17
chuckebaby
Max Output Level: 0 dBFS
  • Total Posts : 13146
  • Joined: 2011/01/04 14:55:28
  • Status: offline
Re:Recording Levels 2012/02/29 13:38:09 (permalink)
no that digigo link is bad here at my side bro...my anti virus is flippin out

Windows 8.1 X64 Sonar Platinum x64
Custom built: Asrock z97 1150 - Intel I7 4790k - 16GB corsair DDR3 1600 - PNY SSD 220GB
Focusrite Saffire 18I8 - Mackie Control
   
#18
batsbrew
Max Output Level: 0 dBFS
  • Total Posts : 10037
  • Joined: 2007/06/07 16:02:32
  • Location: SL,UT
  • Status: offline
Re:Recording Levels 2012/02/29 13:49:47 (permalink)
works fine for me.



let's try this:


Level Practices (Part 2) (Includes the K-System) 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 integratedsystem of metering and monitoring that will encourage more consistent leveling practices among the three disciplines. This system handles the issue of differing dynamic range requirements far more elegantly and ergonomically than in the past. We're on the threshold of the introduction of a new, high-resolution consumer audio format and we have a unique opportunity to implement a 21st Century approach to leveling, that integrates with the concept of Metadata. Let's try to make this a worldwide standard to leave a legacy of better recordings in the 21st Century. 

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

Summary of VU Inconsistencies and Errors

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

Ballistics
The meter's ballistics were designed to "look good" with spoken word. Its 300 ms integration time gives it a syllabic response, which looks very "comfortable" with speech, but doesn't make it accurate. One time constant cannot sum up the complex multiple time constants required to model the loudness perception of the human listener. Skilled users soon learned that an occasional short "burst" from 0 to +3 VU would probably not cause distortion, and usually was meaningless as far as a loudness change.
Scale
In 1939, logarithmic amplifiers were large and cumbersome to construct, and it was desirable to use a simple passive circuit. The result is a meter where every decibel of change is not given equal merit. The top 50% of the physical scale is devoted to only the top 6 dB of dynamic range, and the meter's useable dynamic range is only about 13 dB. Not realizing this fundamental fact, inexperienced and experienced operators alike tend to push audio levels and/or compress them to stay within this visible range. With uncompressed material, the needle fluctuates far greater than the perceived loudness change and it is difficult to distinguish compressed from uncompressed material by the meter. Soft material may hardly move the meter, but be well within the acceptable limits for the medium and the intended listening environment.

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

Lack of conformance to standards
There are large numbers of improperly-terminated mechanical VU meters and inexpensively-constructed indicators which are labelled "VU" in current use. These disparate meters contribute to disagreements among program producers reading different instruments. A true VU meter is a rather expensive device. It's not a VU meter unless it meets the standard.
Over the past 60 years, psychoacousticians have learned how to measure perceived loudness much better than a VU. Despite all these facts, the VU meter is a very primitive loudness meter. In addition, current digital technology permits us to easily correct the non-linear scale, its dynamic range, ballistics,and frequency response.
II. Current-day levelling problems

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

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

Is this what will happen to the next generation carrier? (e.g. DVD-A, SACD). It will, if we don't take steps to stop it. Unlike with the LP, there is no PHYSICAL limit to the average level we can place on a digital medium. Note that there is a point of diminishing returns above about -14 dBFS. Dynamic inversion begins to occur and the program material usually stops sounding louder because it loses clarity and transient response.
III. The Magic of "83" with Film Mixes
In the music world, everyone currently determines their own average record level, and adjusts their monitor accordingly. With no standard, subjective loudness varies from CD to CD in popular music as much as 10-12 dB, which is unacceptable by any professional standard. But in the film world, films are consistent from one to another, because the monitoring gain has been standardized. In 1983, as workshops chairman of the AES Convention, I invited Tomlinson Holman of Lucasfilm to demonstrate the sound techniques used in creating the Star Wars films. Dolby systems engineers labored for two days to calibrate the reproduction system in New York's flagship Ziegfeld theatre. Over 1000 convention attendees filled the theatre center section. At the end of the demonstration, Tom asked for a show of hands. "How many of you thought the sound was too loud?" About four hands were raised. "How many thought it was too soft?" No hands. "How many thought it was just right?" At least 996 audio engineers raised their hands.

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

When AC-3 and DTS became available for home theatre, many authorities recommended lowering the monitor gain by 6 dB because a typical home listening room does not accomodate high SPLs and wide dynamic range. If a DVD contains the wide range theatre mix, many home listeners complain that "this DVD is too loud", or "I lose the dialogue when I turn the volume down so that the effects don't blast." With reduced monitor gain, the soft passages become too soft. For such listeners, the dynamic range may have to be reduced by 6 dB (6 dB upward Compression, or dynamic range reduction) in order to use less monitor gain. 
Metadata are coded data which contain information about signal dynamics and intended loudness; this will resolve the conflict between listeners who want the full theatrical experience and those who need to listen softly. But without metadata there are only two solutions: a) to compromise the audio soundtrack by compressing it, or better, b) use an optional compressor for the home system. With the later approach the source audio is uncompromised.
IV. The Magic of "-6 dB" Monitor Gain for the Home
In the 21st century, home theatre, music, and computers are becoming united. Many, if not most, consumers will eventually be auditioning music discs on the same system that plays broadcast television, home theatre (DVDs), and possibly even web-audio, e.g. MP3. Music-only discs are often used as casual or background music, but I am specifically referring to foreground music that the discerning consumer or audiophile will play at normal or full "enjoyment" loudness.

With the integration of media into a single system, it is in the direct interest of music producers to think holistically and unite with video and film producers for a more consistent consumer audio presentation. Music producers experimenting with 5.1 surround must pay more than casual attention to monitor level calibration. They have already discovered the annoyance that a typical pop CD will blast the sound system when inserted into a DVD player after a movie has been played. Recently a DVD and soundtrack CD were produced of the classic rock music movie Yellow Submarine. Reviewers complained that the CD is much louder and less dynamic than the DVD. Audio CDs should not be degraded for the sake of a "loudness competition". CDs can and should be produced to the same audio quality standard as the DVD.
New program producers with little experience in audio production are coming into the audio field from the computer, software and computer games arena. We are entering an era where the learning curve is high, engineer's experience is low, and the monitors they use to make program judgments are less than ideal. It is our responsibility to educate engineers on how to make loudness judgments. A plethora of peak-only meters on every computer, DAT machine and digital console do not provide information on program loudness. Engineers must learn that the sole purpose of the peak meter is to protect the medium and that something more like average level affects the program's loudness. Bear in mind that the bandwidth and frequency distribution of the signal also affect program loudness.

As a music mastering engineer, I have been studying the perceived loudness of music compact discs for over 15 years. Around 1993, I installed a 1 dB/per step monitor control for repeatability. In an effort to achieve greater consistency from disc to disc, I made it a point to try to set the monitor gain first, and then master the disc to work well at that monitor gain.
In 1996, we measured that monitor gain, and found it to be 6 dB less than the film-standard for most of the pop music we were mastering. To calibrate a monitor to the film-standard, play a standardized pink noise calibration signal whose amplitude is -20 dB FS RMS, on one channel (loudspeaker) at a time. Adjust the monitor gain to yield 83 dB SPL using a meter with C-weighted, slow response. Call this gain 0 dB, the reference, and you will find the pop-music "standard" monitor gain at 6 dB below this reference. 
By now, we've mastered hundreds of pop CDs working at monitor gain 6 dB below the reference, with very satisfied clients. However, if monitor gain is further reduced, average recorded level tends go up because the mastering engineer seeks the same loudness to the ears. Since the average program level is now closer to the maximum permissible peak level, more compression/limiting must be used to keep the system from overloading. Increased compression/limiting is potentially damaging to the program material, resulting in a distorted, crowded, unnatural sound. Clients must be informed that they can't get something for nothing; a hotter record means lower sound quality.

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

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

As the years went on, trying to "hold the fort", I gradually raised the average level of mastered CDs only when requested, which forced the monitor gain to be reduced from 1 to several dB. For every decibel of increased average level, considerably more damage is done to the sound. We often note severe processor distortion when the monitor gain falls below -6 dB. Consumers find their volume controls at the bottom of their travel, where a small control movement produces awkward level changes.
V. The relationship between SPL and 0 VU
In 1994, I installed a pair of Dorrough meters, in order to view the average and peak level simultaneously on the same scale. These meters use a scale with 0 "average" (a quasi-VU characteristic I'll call "AVG") placed at 14 dB below full digital scale, and full scale marked as +14 dB. Music mastering engineers often use this scale, since a typical stereo 1/2" 30 IPS analog tape has approximately 14 dB headroom above 0 VU.

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

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

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

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

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

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

The peak and average scales are calibrated as per AES-17, so that peak and average sections are referenced to the same decibel value with a sine wave signal. In other words, +20 dB RMS with sine wave reads the same as +20 dB peak, and this parity will be true only with a sine wave. Analog voltage level is not specified in the K-system, only SPL and digital values. There is no conflict with -18 dBFS analog reference points commonly used in Europe.
VII. Production Techniques with the K-System
To use the system, first choose one of the three meters based on the intended application. Wide dynamic range material probably requires K-20 and medium range material K-14. Then, calibrate the monitor gain where 0dB on the meter yields 83 dB SPL (per channel, C-Weighted, slow speed). 0dB always represents the same calibrated SPL on all three scales, unifying production practices worldwide. The K-system is not just a meter scale, it is an integrated system tied to monitoring gain.

A manual for a certain digital limiter reads: "For best results, start out with a threshold of -6 dB FS". This is like saying "always put a teaspoon of salt and pepper on your food before tasting it." This kind of bad advice does not encourage proper production practice. A gain reduction meter is not an indication of loudness. Proper metering and monitoring practice is the only solution.
If console and workstation designers standardize on the K-System it will make it easier for engineers to move programs from studio to studio. Sound quality will improve by uniting the steps of pre-production (recording and mixing), post-production (mastering) and metadata (authoring) with a common "level" language. By anchoring operations to a consistent monitor reference, operators will produce more consistent output, and everyone will recognize what the meter means.
If making an audiophile recording, then use K-20, if making "typical" pop or rock music, or audio for video, then probably choose K-14. K-12 should be reserved strictly for audio to be dedicated to broadcast; broadcast recording engineers may certainly choose K-14 if they feel it fits their program material. Pop engineers are encouraged to use K-20 when the music has useful dynamic range.
The two prime scales, K-20 and K-14, will create a cluster near two different monitor gain positions. People who listen to both classical and popular music are already used to moving their monitor gains about 6 dB (sometimes 8 to 12 dB with the hottest pop CDs). It will become a joy to find that only two monitor positions satisfy most production chores. With care, producers can reduce program differences even further by ignoring the meter for the most part, and working solely with the calibrated monitor.

Using the Meter's Red Zone. This 88-90 dB+ region is used in films for explosions and special effects. In music recording, naturally-recorded (uncompressed) large symphonic ensembles and big bands reach +3 to +4 dB on the average scale on the loudest (fortissimo) passages. Rock and electric pop music take advantage of this "loud zone", since climaxes, loud choruses and occasional peak moments sound incorrect if they only reach 0dB (forte) on any K-system meter. Composers have equated fortissimo to 88-90+ dB since the time of Beethoven. Use this range occasionally, otherwise it is musically incorrect (and ear-damaging). If engineers find themselves using the red zone all the time, then either the monitor gain is not properly calibrated, the music is extremely unusual (e.g. "heavy metal"), or the engineer needs more monitor gain to correlate with his or her personal sensitivities. Otherwise the recording will end up overcompressed, with squashed transients, and its loudness quotient out of line with K-System guidelines.
Equal Loudness Contours
Mastering engineers are more inclined to work with a constant monitor gain. But many music mixing engineers work at a much higher SPL, and also vary their monitor gain to check the mix at different SPLs. I recommend that mix engineers calibrate your monitor attenuators so you can always return to the recommended standard for the majority of the mix. Otherwise it is likely the mix will not translate to other venues, since the equal-loudness contours indicate a program will be bass-shy when reproduced at a lower (normal) level.
Tracking/Mixing/Mastering
The K-System will probably not be needed for multitracking--a simple peak meter is probably sufficient. For highest sound quality, use K-20 while mixing and save K-14 for the calibrated mastering suite. If mixing to analog tape, work at K-20, and realize that the peak levels off tape will not exceed about +14. K-20 doesn't prevent the mix engineer from using compressors during mixing, but the author hopes that engineers will return towards using compression as an esthetic device rather than a "loudness-maker."
Using K-20 during mix encourages a clean-sounding mix that's advantageous to the mastering engineer. At that point, the producer and mastering engineer should discuss whether the program should be converted to K-14, or remain at K-20. The K-System can become the lingua franca of interchange within the industry, avoiding the current problem where different mix engineers work on parts of an album to different standards of loudness and compression.

When the K-System is not available
Current-day analog mixing consoles equipped with VUs are far less of a problem than digital models with only peak meters. Calibrate the mixdown A/D gain to -20 dBFS at 0 VU, and mix normally with the analog console and VUs. However, mixing consoles should be retro fitted with calibrated monitor attenuators so the mix engineer can repeatably return to the same monitor setting.
Compression is a powerful esthetic tool. But with higher monitor gain, less compression is needed to make material sound good or "punchy." For pop music, many K-14 presentations sound better than K-20, with skillfully-applied dynamics processing by a mastering engineer working in a calibrated room. But clearly, the higher the K-number, the easier it is to make it sound "open" and clean. Use monitor systems with good headroom so that monitor compression does not contaminate the judgment of program transients.
Adapting large theatre material to home use may require a change of monitor gain and meter scale. Producers may choose to compress the original 6-channel theatre master, or better, remix the entire program from the multi-track stems (submixes). With care, most of the virtues and impact of the original production can be maintained in the home. Even audiophiles will find a well-mastered K-14 program to be enjoyable and dynamic. It is desirable to try to fit this reduced-range mix on the same DVD as the wide-range theatre mix.
Multichannel to Stereo Reductions
The current legacy of loud pop CDs creates a dilemma because DVD players can also play CDs. Producers should try to create the 5.1 mix of a project at K-20. If possible, the stereo version should also be mixed and mastered at K-20. While a K-20 CD will not be as loud as many current pop CDs, it may be more dynamic and enjoyable, and there will not be a serious loudness jump compared to K-20 DVDs in the same player. If the producer insists on a "louder" CD, try to make it no louder than K-14, in which case there will only be 6 dB loudness difference between the DVD and the audio CD. Tell the producer that the vast majority of great-sounding pop CDs have been made at K-14 and the CD will be consistent with the lot, even if it isn't as hot as the current hypercompressed "fashion." It's the hypercompressed CD that's out of line, not the K-14.
Full scale peaks and SNR
It is a common myth that audible signal-to-noise ratio will deteriorate if a recording does not reach full scale digital. On the contrary, the actual loudness of the program determines the program's perceived signal-to-noise ratio. The position of the listener's monitor level control determines the perceived loudness of the system noise. If two similar music programs reach 0 on the K-system's average meter, even if one peaks to full scale and the other does not, both programs will have similar perceived SNR. Especially with 20-24 bit converters, the mix does not have to reach full scale (peak). Use the averaging meter and your ears as you normally would, and with K-20, even if the peaks don't hit the top, the mixdown is still considered normal and ready for mastering, with no audible loss of SNR.
Multipurpose Control Rooms
With the K-System, multipurpose production facilities will be able to work with wide-dynamic range productions (music,videos, films) one day, and mix pop music the next. A simultaneous meter scale and monitor gain change accomplishes the job. It seems intuitive to automatically change the meter scale with the monitor gain, but this makes it difficult to illustrate to engineers that K-14 really is louder than K-20.
A simple 1 dB per step monitor attenuator can be constructed, and the operator must shift the meter scale manually.

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

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

Individuals who decide to use a different monitor gain should log it on the tape (file) box, and try to use this point consistently. Even with slight deviations from the recommended K(N) practice, the music world will be far more consistent than the current chaos. Everyone should know the monitor gain they like to use.
At left is a picture of an actual K-14/RMS Meter in operation at the Digital Domain studio, as implemented by Metric Halo labs in the programSpectrafoo for the Macintosh. Spectrafoo versions 3f17 and above include full K-System support and a calibrated RMS pink noise generator. Other meters that conform exactly with K-System guidelines have been implemented by Pinguin for PC, RME in their Digichek software, and Roger Nichols Digital (formerly Elemental audio) Inspector XL. The Dorrough and DK meters nearly meet K-System guidelines but an external RMS meter must be used for pink noise calibration since they use a different type of averaging. In practice with program material, the difference between RMS and other averaging methods is insignificant, especially when you consider that neither method is close enough to a true loudness meter. As of this date, 12/05/07, we are still awaiting a company that will implement the K-System with a loudness characteristic, such as Zwicker.
Audio Cassette Duplication
Cassette duplication has been practiced more as an art than a science, but it should be possible to do better. The K-System may finally put us all on the same page (just in time for obsolescence of the cassette format). It's been difficult for mastering engineers to communicate with audio cassette duplicators, finding a reference level we all can understand. A knowledgeable duplicator once explained that the tape most commonly used cannot tolerate average levels greater than +3 over 185 nW/m (especially at low frequencies) and high frequency peaks greater than about +5-6 are bound to be distorted and/or attenuated. Displaying crest factor makes it easy to identify potential problems; also an engineer can apply cassette high-frequency preemphasis to the meter. Armed with that information, an engineer can make a good cassette master by using a "predistortion" filter with gentle high-frequency compression and equalization. Meter with K-14 or K-20, and put test tone at the K-System reference 0 on the digital master. Peaks must not reach full scale or the cassette will distort. Apparent loudness will be less than the K-standard, but this is a special case.
Classical music
It's hard to get out of the habit of peaking our recordings to the highest permissible level, even though 24-bit systems have a theoretically 48 dB better signal-to-dither-ratio than 16-bit. It is much better for the consumer to have a consistent monitor gain than to peak every recording to full scale digital. I believe that attentive listeners prefer auditioning at or near the natural sound pressure of the original classical ensemble (see Footnote). The dilemma is that string quartets and Renaissance music, among other forms, have low crest factors as well as low natural loudness. Consequently, the string quartet will sound (unnaturally) much louder than the symphony if both are peaked to full scale digital.
I recommend that classical engineers mix by the calibrated monitor, and use the average section of the K-meter only as a guide. It's best to fix the monitor gain at 83 dB and always use the K-20 meter even if the peak level does not reach full scale. There will be less monitoring chaos and more satisfied listeners. However, some classical producers are concerned about loss of resolution in the 16-bit medium and may wish to peak all recordings to full scale. I hope you will reconsider this thought with 24 bit media or SACD.
Narrow Dynamic Range Pop Music
We can avoid a new loudness race and consequent quality reduction if we unite behind the K-System before we start fresh with high-resolution audio media such as DVD-A and SACD. Similar to the above classical music example, pop music with a crest factor much less than 14 dB should not be mastered to peak to full scale, as it will sound too loud.
Recommended:
1: Author with metadata to benefit consumers using equipment that supports metadata
2: If possible, master such discs at K-14
3: Legacy music, remasters from often overcompressed CD material should be reexamined for its loudness character. If possible, reduce the gain during remastering so the average level falls within K-14 guidelines. Even better, remaster the music from unprocessed mixes to undo some of the unnecessary damage incurred during the years of chaos. Some mastering engineers already have made archives without severe processing.


VIII. An Extendable System

Since the K-System is extendable to future methods of measuring loudness, program producers should mark their tape boxes or digital files with an indication which K-meter and monitor calibration was used. For example, "K-14/RMS," or "K-20/Zwicker." I hope that these labels will someday become as common as listings of nanowebers per meter and test tones for analog tapes. If a non-standard monitor gain was used, note that fact on the tape box to aid in post-production authoring and insertion of metadata.
IX. Metadata and the K-System
Dolby AC-3, MPEG2, AAC, and hopefully MLP will take advantage of metadata control words. Pre-production with the K-System will speed the authoring of metadata for broadcast and digital media. Music producers must familiarize themselves with how metadata affects the listening experience. First we'll summarize how the control word Dialnorm is used in digital television. Then we will examine how to take advantage of Dialnorm and MixLevel for music-only productions.
Dialnorm
Dialogue normalization, is used in digital television and radio as "ecumenical gain-riding". Program level is controlled at the decoder, producing a consistent average loudness from program to program; with the amount of attenuation individually calculated for each program. The receiver decodes the dialnorm control word and attenuates the level by the calculated amount, resulting in the "table radio in the kitchen" effect. In an unnatural manner, average levels of sports broadcasts, rock and roll, newscasts, commercials, quiet dramas, soap operas, and classical music all end up at the loudness of average spoken dialogue.
With Dialnorm, the average loudness of all material is reduced to a value of -31 dB FS (LEQ-A). Theatrical films with dialogue at around -27 dB FS will be reduced 4 dB. -31 corresponds not with musical forte, but rather mezzo-piano. For example, a piece of rock and roll, normally meant to be reproduced forte, may be reduced 10 or more dB, while a string quartet may only be reduced 4-5 dB at the decoder. The dialnorm value for a symphony should probably be determined during the second or third movement, or the results will be seriously skewed. We do want the forte passages to be louder than the spoken word! Rock and roll, with its more limited dynamic range, will be attenuated farther from "real life" than the symphony. However, unlike the analog approach, the listener can turn up his receiver gain and experience the original program loudness--without the noise modulation and squashing of current analog broadcast techniques. Or, the listener can choose to turn off dialnorm (on some receivers) and experience a large loudness variance from program to program.
Each program is transmitted with its full intended dynamic range, without any of the compression used in analog broadcasting--the listener will hear the full range of the studio mix. For example, in variety shows, the music group will sound pleasingly louder than the presenter. Crowd noises in sports broadcasts will be excitingly loud, and the announcer's mike will no longer "step on" the effects, because the bus compressor will be banished from the broadcast chain.

Mixlev
Dialnorm does not reproduce the dyamic range of real life from program to program. This is where the optional control word mixlev (mix level) enters the picture. The dialnorm control word is designed for casual listeners, and mixlev for audiophiles or producers. Very simply, mixlev sets the listener's monitor gain to reproduce the SPL used by the original music producer. Only certain critical listeners will be interested in mixlev. If the K-system was used to produce the program, then K-14 material will require a 6 dB reduction in monitor gain compared to K-20, and so on. Mixlev will permit this change to happen automatically and unattended. Attentive listeners using mixlev will no longer have to turn down monitor gains for string quartets, or up for the symphony or (some) rock and roll.
The use of dialnorm and mixlev can be extended to other encoded media, such as DVD-A. Proper application of dialnorm and mixlev, in conjunction with the K-System for pre-production practice--will result in a far more enjoyable and musical experience than we currently have at the end of the 20th century of audio.
X. In Conclusion
Let's bring audio into the 21st century. The K-system is the first integrated approach to monitoring, levelling practices, metering and metadata.

B: Multichannel
There's good news for audio quality: 5.1 surround sound. Current mixes of popular music that I have listened to in 5.1 sound open, clear, beautiful, yet also impacting. I've done meter measurements and listening to a few excellent 20 and 24 bit 5.1 mixes, and they all fall perfectly into the K-20 Standard. Monitor gain ran from 0 dB to -3 dB, mostly depending on taste, as it was perfectly comfortable to listen to all of these particular recordings at 0 dB (reference RP 200).
What became clear while watching the K-20 meter is that the best engineers are using the peak capability of the 5.1 system strictly for headroom. It is possible that I didn't see a single peak to full scale (+20 on the K-20 Meter) on any of these mixes. The averaging portion of the meter operated just as in my recommendations, with occasional peaks to +4 on some of the channels.
Monitor calibration made on an individual speaker basis worked extremely well, with the headroom in each individual channel tending to go up as the number of channels increases. This is simply not a problem with 24 bit (or even 20 bit) recording. System hiss is not evident at RP 200 monitor gains with long-wordlength recording, good D/A converters, modern preamps and power amplifiers.
Another question is: Should we have an overall meter calibrated to a total SPL? If so, what should that SPL be? My initial reactions are that an overall meter is not necessary, at least in mix situations where mix engineers use calibrated monitoring and monitors with good headroom.
Another positive thought. I've been giving 5.1 seminars sponsored by TC, Dynaudio, and DK Meters. To begin the show, I played two stereo masters that I had mastered, and demonstrated some very sophisticated techniques to bump them up (transparently) to 5.1. This is a growing field, and you'll see increasing techniques for doing this, especially when the record company wants a DVD or DVD-A remaster without (horrors) having to pay for a remix.
The good news is I found that the true 5.1 mixes by George Massenburg and others that I was demonstrating sounded so OPEN and clear and beautiful that even I was embarrassed to start from a 24-bit version of my own two masters. I had to remaster the two pieces with about 2 to 4 dB LESS LIMITING in order to make them COMPETE SONICALLY with the 5.1 stuff!!! "Louder is better" just doesn't work when you're in the presence of great masters.
That's right, I predict that the critical mastering engineers of the future will be so embarrassed by the sound quality of the good 5.1 stuff that they won't be able to get away with smashing 5.1 masters. And, hopefully, the two-track reductions that they also remaster (the CD versions) especially if there is a CD layer on the same disc, will be mastered to work at the same LOUDNESS.
In fact, if you tried to turn 5.1 Lyle Lovett, Michael Jackson, Aaron Neville, or Sting into a K-14, they just would sound horrid, on any reasonable 5.1 playback system!
The DK meters, set to K-20 demonstrated clearly that K-20 rules in 5.1. In fact, after a while I simply turned off the peak portion of the meter as it was distracting. So we could watch the VU-style levels and see the techniques used by each of the mix engineers. At K-20 and with 6 speakers running, you have so much headroom that it is hardly necessary to watch the peak meters at all. Furthermore, at 24 bits, there is absolutely no necessity to hit 0 dBFS ANYMORE AT ALL.
The proof is in the pudding, when you try your first 5.1 master you will see clearly what I mean. K-20-style metering and calibrated monitoring becomes a MUST in 5.1.
If you are interested in discussing the ramifications of these topics, please contact the author, Bob Katz.

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

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


Bats Brew music Streaming
Bats Brew albums:
"Trouble"
"Stay"
"The Time is Magic"
--
Sonar 6 PE>Bandlab Cakewalk>Studio One 3.5>RME BFP>i7-7700 3.6GHz>MSI B250M>G.Skill Ripjaws 4 series 16GB>Samsung 960 EVO m.2ssd>W 10 Pro
 
#19
Jeff Evans
Max Output Level: -24 dBFS
  • Total Posts : 5139
  • Joined: 2009/04/13 18:20:16
  • Location: Ballarat, Australia
  • Status: offline
Re:Recording Levels 2012/02/29 15:27:15 (permalink)
People are only referring to peak levels when they mention any recording levels, and that does not take into account what the rms level of the signal is underneath it.

The K system keeps the rms levels on tracks and busses constant and can do it at 3 different digital reference levels eg -20, -14, -12. Peak levels will then vary and they take care of themselves most of the time. We had RMS metering back in the analog days but it seems to have been lost or pushed aside in favour of peak metering instead. Both are the best option. The K system also ensures that while you have rms values around your chosen digital reference level, the actual metering is high up on the meter and showing around 0 db VU which is where you need to see it the most.

DAW's that do have rms metering have it way down too low on the scale (because it is down there) and it is not useful.

These VU meters are very good and the closest thing that I have tested for a while in the VST VU meter area:

http://www.klanghelm.com/

(If those compressors that are advertised on that site are as good as their VU meters they will be fantastic!) The BlueCat meter also works well too.

All you have to do is select a particular K System ref level and work to it. Use the VU's on tracks and busses, tell them what the ref level is, simple as that. A VU meter showing you the rms level on a track or buss makes it easy to set the correct recording levels in your system and you won't clip it very often either. You still keep a watchful eye on your peak meters as well to ensure that very transient or short fast sounds that won't move the VU much will still not clip your system.

I had to fine tune the settings on the Klang meters before I got them moving in perfect harmony with my expensive hardware API meters. If anyone gets these and wants these settings I am happy to publish the settings in a post or PM then to you.

Using the K System goes a long way to all your tracks and busses being at a constant and correct level right from the start. Take your time setting incoming record levels. People rush this and get it wrong. Monitor over a wide range of performance dynamics to get the best setting for your input gain. (VU's make it easy) After tracking correctly there will be no processing (normalising) and/or trimming required later because your levels are already right! Your mixes will also be much more even in their rms levels and mastering is also way easier when your mixes are like this.

K System is also about keeping your monitoring level in your room constant no matter what the K system ref level is. (Very important!) You mix and master better simply by doing this alone.




post edited by Jeff Evans - 2012/02/29 15:30:22

Specs i5-2500K 3.5 Ghz - 8 Gb RAM - Win 7 64 bit - ATI Radeon HD6900 Series - RME PCI HDSP9632 - Steinberg Midex 8 Midi interface - Faderport 8- Studio One V4 - iMac 2.5Ghz Core i5 - Sierra 10.12.6 - Focusrite Clarett thunderbolt interface 
 
Poor minds talk about people, average minds talk about events, great minds talk about ideas -Eleanor Roosevelt
#20
ChuckC
Max Output Level: -61 dBFS
  • Total Posts : 1488
  • Joined: 2010/02/13 01:22:55
  • Location: Port Charlotte, Fl
  • Status: offline
Re:Recording Levels 2012/02/29 16:07:54 (permalink) ☄ Helpful
When I record I generally try to keep the levels at -6 /-8 db and make absoulutely sure there is no clipping.  When I start mixing I pull up voxengo span and set the k metering to k-14  and reference it frequently. When mastering I do the same till I am done.  Then I play the whole track start to finish, if I peak at say -.9 I move the master fader up by .8 so that my highest peak is at -.1 This gives me a little higher RMS and percieved loudness without limiting the piss out of it.   Just my take and what I have learned to do so far with pretty good results.

ADK Built DAW, W7, Sonar Platinum, Studio One Pro,Yamaha HS8's & HS8S  Presonus Studio/Live 24.4.2, A few decent mic pre's,  lots of mics, 57's,58 betas, Sm7b, LD Condensors, Small condensors, Senn 421's,  DI's,  Sans Amp, A few guitar amps etc. Guitars : Gib. LP, Epi. Lp, Dillion Tele, Ibanez beater, Ibanez Ergodyne 4 String bass, Mapex Mars series 6 pc. studio kit, cymbals and other sh*t.
http://www.everythingiam.net/
http://www.stormroomstudios.com
Some of my productions: http://soundcloud.com/stormroomstudios
#21
droddey
Max Output Level: -24 dBFS
  • Total Posts : 5147
  • Joined: 2007/02/09 03:44:49
  • Location: Mountain View, CA
  • Status: offline
Re:Recording Levels 2012/02/29 16:12:10 (permalink)
You can of course do a 'poor mans' k-system setup and get most of the advantages. Basically just do something like this:

1. Set your analog volume controller knob at some reasonable level, like 3/4s up, and mark that position.
2. Select a number of commercial tracks that represent a wide range of compression. So something like maybe Dark Side of the Moon on the low end and something like Green Day on the high end.
3. Import these songs onto tracks. Leave the track faders at 0
4. Pull the master fader down until those tracks are peaking at whatever you want your reference level to be, say -12dBFS on the master bus.
5. Now start playing these tracks. Adjust your speaker sensitivity levels so that the most compressed ones are uncomfortably loud, and the least compressed ones are a little bit quiet. That will mean that reasonably compressed stuff will be somewhere in the middle. Go back and forthe between the tracks to find a happy medium, but be sure that the heavily compressed stuff is just too damn loud to listen to comfortably.


So now what you have is a point on the analog volume control that you can go to and know that it's at a known point, and you have a master volume peak level (-12dBFS in this case) that you can then allow your own stuff to peak at (with the master fader back up to 0dBFS again of course.) So now, since you've set a known master bus peak, the only way you can get the music louder is to compress it more (bring up the average level.) And you have already determined that the actual SPL in the room is set such that overly compressed music is too loud.

So basically you have a situation now where you will tend to naturally gravitate towards a reasonable and appropriate level of compression for the song you are working on. Too little and it starts getting a bit too quiet. Too much and it starts getting uncomfortably loud.

This is not a formal setup, and doesn't use an SPL meter which is convenient to have, but it actually works very well and if you later do a real one you'll probably find out that you end up very close to the same thing. The basically point of it is to just force you towards appropriate levels of compression naturally, and of course it also provides you with a standard monitoring level so that you can have a consistent Fletcher/Munson response for gauging the overall mids vs. lows/highs balance of the mix (which will change a lot as the volume in the room goes up and down.)

If you have an SPL meter, you can measure the commercial tracks' SPL in the room, then put the master bus fader back up to 0dBFS and turn down the analog volume knob until you get back to the same SPL. Mark this one and you now have a reference position for listening to commercial music that should be at the same SPL as your mix levels are, which is nice for comparison purposes.

Dean Roddey
Chairman/CTO, Charmed Quark Systems
www.charmedquark.com
#22
batsbrew
Max Output Level: 0 dBFS
  • Total Posts : 10037
  • Joined: 2007/06/07 16:02:32
  • Location: SL,UT
  • Status: offline
Re:Recording Levels 2012/02/29 16:35:20 (permalink)
People are only referring to peak levels when they mention any recording levels, and that does not take into account what the rms level of the signal is underneath it.  



This is incorrect.


please note my last post:


so, back on topic: 

i personally record almost all of my tracks with no more than -10db peaks..... 

with the exception of drums, which i allow to -8db peaks, but i process them at mixdown, and knock the peaks off with compressors and limiters.... 


i find the closer i get to -8db peak as a MAXIMUM allowable peak.... 
the more 'crunchy' the sound is. 

that actually works for me, with drums, but not with anything else. 


if i record something all the way down to -14db peak (which means an average RMS value of about -23), that seems to be the "SWEET SPOT" for my system, signal chain, method of recording, etc. 



Bats Brew music Streaming
Bats Brew albums:
"Trouble"
"Stay"
"The Time is Magic"
--
Sonar 6 PE>Bandlab Cakewalk>Studio One 3.5>RME BFP>i7-7700 3.6GHz>MSI B250M>G.Skill Ripjaws 4 series 16GB>Samsung 960 EVO m.2ssd>W 10 Pro
 
#23
Jeff Evans
Max Output Level: -24 dBFS
  • Total Posts : 5139
  • Joined: 2009/04/13 18:20:16
  • Location: Ballarat, Australia
  • Status: offline
Re:Recording Levels 2012/02/29 18:45:32 (permalink)
Most people are only seeing and therefore working with peak levels only and that is the reasoning behind my statement as such. Few are measuring rms levels consistently and accurately.

In batsbrew case above he is implying that his rms level is down at -23 and any transient associated with that sound is up at -14 meaning the transient is 9db above the rms value. (this would be for a transient type of sound with some degree of attack) K System wise this is close to the -20 situation. I would adjust the rms signal to be around there putting the peak at -11 above that. That is the good thing about the K-20 ref level. There is 20 db of headroom above it so that peak still has 11 db to go before it clips anything.

I don't agree that digital recording suddenly has a sweet spot. I feel it sounds very similar over a wide variety of levels. Especially at 24 bit recording where this is much less of an issue.

If I were recording bats  signal now at a K-14 db level then the rms part would be at -14 and the peak would extend up to -5db which is still fine but with only 5 db of headroom above it. The K-14 ref levels is going to 6db louder than the k-20 ref level but with less headroom for emergencies etc..(Note with other types signals you can have an rms part still at K-14 and the peaks only 2 or 3 db avove that)

There is a simpler way to calibrate DAW's and monitor systems as well. You do need some measuring devices you cannot get around it. Load up a stereo pink noise test signal at the desired K syetem ref level. (Bob Katz webiste has a perfect pink noise test signal at K-20 if you cannot make it yourself, just add 6 and 8 db to it to make the K-14 and K-12 test signals respectively) Your VU's should be showing very close to 0db VU. (They have to be set for the same ref level obviously, Klang meters can bet set at any ref level)

You simply adjust your monitor gain for either 83 db out of each seaker alone or 85 db for both. (Pink noise is a must otherwise standing waves will severly effect tones by only moving the meter slightly)

If you are comparing commercially mastered music to your own work at your DAW ref level etc it will obviously be much louder and pin the VU's. So you simply park that on a track and drop it down so the VU's show 0db now.  eg will be -6 to -7db on average.

When you master now you simply put the commercial track back to unity gain on the faders and adjust your VU meters to be calibrated at say -6 or -7db rms instead. Then the commercial trrack will appear normal and peak 0VU and now your DAW output will be shy by the same amount. You perform your mastering processing and slowly get it up to match. The monitor gain in your room has to be lowered by the same amount as well to maintain the same 85 db in the room.

Specs i5-2500K 3.5 Ghz - 8 Gb RAM - Win 7 64 bit - ATI Radeon HD6900 Series - RME PCI HDSP9632 - Steinberg Midex 8 Midi interface - Faderport 8- Studio One V4 - iMac 2.5Ghz Core i5 - Sierra 10.12.6 - Focusrite Clarett thunderbolt interface 
 
Poor minds talk about people, average minds talk about events, great minds talk about ideas -Eleanor Roosevelt
#24
droddey
Max Output Level: -24 dBFS
  • Total Posts : 5147
  • Joined: 2007/02/09 03:44:49
  • Location: Mountain View, CA
  • Status: offline
Re:Recording Levels 2012/02/29 19:01:30 (permalink)
83dB can be too loud for a small room. It definitely was in mine. The ultimate purpose of it isn't to reach a particular SPL, but to have overly compressed material be too loud in your room, reasonably compressed material be just right, and very uncompressed material will probably be a bit overly quiet in the quiet parts. In a studio sized room, 83dB is probably there. In a small apartment room, I just felt it was too loud. I went more for the mid-70s. Going back and doing it by ear like I did above (as an experiement) got me to almost exactly the same place, which isn't suprising.

Obviously if you have the SPL meter then use it. But it's not a requirement that you use 83dB or 85dB. If that's too loud, then it's too loud and adjust downwards.

Dean Roddey
Chairman/CTO, Charmed Quark Systems
www.charmedquark.com
#25
Jeff Evans
Max Output Level: -24 dBFS
  • Total Posts : 5139
  • Joined: 2009/04/13 18:20:16
  • Location: Ballarat, Australia
  • Status: offline
Re:Recording Levels 2012/02/29 19:15:54 (permalink)
I think I may have mentioned this once before. 85 dbA is loud. 85 dbC weighting is not very loud. It is a beautiful volume. Listen to it all day long, no fatigue. C Weighting is the correct weighting to use.

PS Dean is right though. Even at C weighting 85 db of pink noise is still quite loud but 85 db of music is not. And we don't listen to pink noise all day do we? So that is why the pink noise signal can seem a little loud.
post edited by Jeff Evans - 2012/02/29 19:19:45

Specs i5-2500K 3.5 Ghz - 8 Gb RAM - Win 7 64 bit - ATI Radeon HD6900 Series - RME PCI HDSP9632 - Steinberg Midex 8 Midi interface - Faderport 8- Studio One V4 - iMac 2.5Ghz Core i5 - Sierra 10.12.6 - Focusrite Clarett thunderbolt interface 
 
Poor minds talk about people, average minds talk about events, great minds talk about ideas -Eleanor Roosevelt
#26
spacealf
Max Output Level: -54 dBFS
  • Total Posts : 2133
  • Joined: 2010/11/18 17:44:34
  • Status: offline
Re:Recording Levels 2012/02/29 20:12:03 (permalink)
All I know is that computers are 100% not over, maybe squeezing 103% at most. Get to 105% and you what I call a A-hole, where the recording peaks over the limit a computer can handle and drops to the other side of the center line putting a cracking noise in the recording (I done this not on purpose with old band analog tapes.) Analog recording could go over the limit but distortion may be a factor. Better studio conditions of digital may handle more, but with home recording equipment and computers that is the way that it works. The VU meters in Sonar can be adjusted to respond faster up to a point if your computer can handle the load. Right now they probably are 40ms delay where you can set them to 10ms delay I think (I am correct) but it will take more CPU power. Whatever happens it is a cumulative effect with each track adding to the total overall volume in the end. Of course some tracks will be softer than recorded in the end while others will have to be lowered to keep from peaking over 100% on a computer. No leeway (unless and probably bigger studios have better equipment that allows them to do a little more, not so with home equipment). The more you pay the more you get, but there is a point where increasing the total production becomes cost prohibitive in the end and may not be needed.
#27
droddey
Max Output Level: -24 dBFS
  • Total Posts : 5147
  • Joined: 2007/02/09 03:44:49
  • Location: Mountain View, CA
  • Status: offline
Re:Recording Levels 2012/02/29 22:38:56 (permalink)
Just an an aside, Gearslutz.com is now coming up as a porn site as well, anyone else seeing that?

Dean Roddey
Chairman/CTO, Charmed Quark Systems
www.charmedquark.com
#28
Philip
Max Output Level: -34.5 dBFS
  • Total Posts : 4062
  • Joined: 2007/03/21 13:09:13
  • Status: offline
Re:Recording Levels 2012/03/01 20:35:07 (permalink)
For my happier mediums and crowded mixes:

Lead vox track: -10 dcbs as a starting point for peaks

Kick: - 4 dcbs (peaks)

Snare: - 6 dcbs

Backing vocs - 20 dcbs

Ambient strings: - 12 dcbs

Guitars: if Haas'd: -15 dcbs

Chuga guitars: -6 dcbs +/-

During mixing, the ears take over and violate a lot of my conventions.

Panning requires unpredictable ear-decisions. 

Extreme panning sometimes requires lower volumes (rock) or higher volumes (dance).

Per Bat, drums dominate my dancier mixes, and are louder.

Philip  
(Isa 5:12 And the harp, and the viol, the tabret, and pipe, and wine, are in their feasts: but they regard not the work of the LORD)

Raised-Again 3http://soundclick.com/share.cfm?id=12307501
#29
Jeff Evans
Max Output Level: -24 dBFS
  • Total Posts : 5139
  • Joined: 2009/04/13 18:20:16
  • Location: Ballarat, Australia
  • Status: offline
Re:Recording Levels 2012/03/01 21:12:32 (permalink)
Hi Phillip Just a few observations. 

What is a dcbs? Is this your slang for decibels. Others like my self may be wondering what unit you are referring to. Decibels are best abbreviated to db. That is the convention.

Are you referring to peak or rms levels. Big difference!

Also you cannot recommend levels for a mix because there are way too many variables. The first is how loud the track is recorded to start with. And the other is what you think is a nice level for say backing vocals may be too loud or too soft for me or others. I think with mix levels people just have to put things where they think they sound best and what suits the material the best.

Specs i5-2500K 3.5 Ghz - 8 Gb RAM - Win 7 64 bit - ATI Radeon HD6900 Series - RME PCI HDSP9632 - Steinberg Midex 8 Midi interface - Faderport 8- Studio One V4 - iMac 2.5Ghz Core i5 - Sierra 10.12.6 - Focusrite Clarett thunderbolt interface 
 
Poor minds talk about people, average minds talk about events, great minds talk about ideas -Eleanor Roosevelt
#30
Page: 12 > Showing page 1 of 2
Jump to:
© 2025 APG vNext Commercial Version 5.1