Why's It So Loud?or Don't You Play any Nice Songs?
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NOTE - make sure you scroll down and read the rant below! I try to explain why this is happening - at least what I think (ya, I know...)
Best quote I've ever heard on the subject from the amazing Bob Katz: "But when all the lights are lit on the peak, you win, right?"
Here are some links to mastering software analysis of part of a song from the latest Metallica CD - Death Magnetic.
NOTE: each page is a bit large. This is a Katz K-14 metering with AES  17 for RMS. Notice the crest factor - the RMS level  to peak levels. The extreme RMS values are some of the highest I've seen in commercial recordings.
I will not offer comments or qualitative opinions of the recordings - I think Rubin may like the controversy esp. in this music biz climate. 

From the CD - ripped into Wavelab 6.01
At this moment in the song, peak is showing 0 DBFS (+14 is K14 0) with an RMS value at 2.85 db below that (+11.15 K14).

From ITUNES - using the ITUNES Burn CD-
Ripped into Wavelab 6.01
At this moment in the song,  peak is showing  0.63 DBFS (+13 where +14 is K14 0) with a RMS only 3 below that (+10.49 K14).
 Now this is interesting - on this page note the difference in selections on the CD - some are brickwalled/flat-topped and some are ragged - in fact, some of the ones flat-topped actually have HIGHER RMS to Peak ratios.

Click here or on the above image - CD rip

note the peak limiting is unlike typical minimal noise shaped brick wall waveforms in which  the top of the waves are totally flat.

Click here or on the above image - ITunes rip

 Again, also note the peak limiting is unlike typical  noise shaped brick wall waveforms in which  the top of the waves are totally flat

Click here or on the above image - CD track compare

Click here or on the above image - CD Track 6 further analysis

Comments to a Lefsetz article:

I really got a buncha meetings for my day job today  so I'd love to explore this a bit more... but hey here's some of my observations and opinions.... for what any of it is worth -

What I think your reader means is the change in crest factor between new stuff and older recordings. In my mastering suite, I have reference CD rips from recent stuff with the typical squashed sound and stuff, even as recent as Blood Sugar Sex Magic and Pearl Jams's 10... big diff in RMS vs Peak ratios....

I believe the reason for the 6db crest factor is the change of the consumer systems... and maybe what Katz (I've talked to him a few times) mentions, a shifting threshold in the consumers hearing/ambient noise factors. Significance must be given though, to the smaller driver surface area, now typical in modern consumer repro systems.

I think a lot of investigation needs to be focused on the comparison of speaker driver surface area, the effects on instantaneous excitation of the acoustic environment, the ratio and linearity of frequency response to this instantaneous ability in typical home acoustic environments, and how the ear perceives the difference in loudness (ie RMS vs peak program) to these factors.

In his book Mastering Audio (he's fucking amazing/great) he delves into the whole squished audio thing....

As to clipping consumer CD players, a few things... recall I design this shit.... the post DAC analog stages vary as to the matching the designer achieves with regard to 0dB-FS levels of the DAC amp (usually an opamp) to the remaining analog stages... I've noticed a significant difference in low-end CD players.

When mastering Jon's stuff, I noticed severe clipping when I ran close to -0.1dB-FS (this is where the 16 bits go to all 1's) on a few of the small CD players in the house. Believe me, when on a small budget like this, you make the rounds thru the house and to every Best Buy, Tweeter, Target with master CD in hand to see how it translates. Any colleague with a CD in the car gets approached...

So I began running absolute noise shaped (this means there's no two- four 16 bit words adjacent in the stream when it gets near the peak limit) to -0.3dB-FS.

Problem solved.... after dissecting a few of the offending units and placing a scope probe on the DAC outputs and the following analog stages, I noticed a huge difference between the CD's limited at -0.1dB-FS and those at -0.3.

Some guys set their limiters at -0.01dB-FS - I've had some major label releases totally trash these poor little CD players; even some of the high end units suffer from this type of overload.

As to my theory, look at most consumer 5.1 systems. Very small driver area. Higher RMS vs peak levels will make most program material sound louder on smaller systems, due to the fact that the ear/brain gauges loudness based on continuous energy levels and has very limited ability to discern peak information.

The majority of the people listen to earbuds.... they're tone deaf... want to bounce to thumpy synth dance music or noisy rock... with sound systems that have a lot less fidelity than the speakers you mentioned you have.

Did you ever listen to the sound coming from the guys that DO spend money on cars stereos?  Yea... probably something like 10% IM distortion. Whumpa Whumpa ... the fucking trunk lid's vibrating. That's real pleasant...great fidelity, even if the source was a zillion bit/1GHz sample rates.

With this in mind, realize that the consumer will usually end up doing their own sort of limiting - - clipping to the power supply rails in their playback system.

I really don't think the majority of people care a lot about "fidelity"...

And it's really not a "new" thing. Just look how long studios, all studios, have had "nearfields/bookshelf speakers" like Yamaha NS-10's (usually with toilet paper taped over the tweeter), and Auratones to mix thru.  Then of course you gotta check your mix in a standard-equipped auto sound system, with the smiley face on the graphic EQ (see Bob Katz's "Mastering Audio 1st Edition - 'Here's what we're up against' " photo on page 81 for that) just like Suzie has her car stereo adjusted. A lot of stuff is mixed/pre-mastered for the lowest common denominator.

With smaller and smaller driver surface area, most music is so squashed to compete and be heard over noisy environs, dynamic range is what? Maybe 6db RMS/Peak?

Just like you mentioned, most people do stuff while they listen, they don't stay in one place as when decent sound reproduction was a novelty, one of few entertainment outlets "on demand" for the consumer. That's one of the advantages for music/aural-only entertainment - you can do other stuff while being entertained. I agree some like you and I may be able to become engrossed in listening to pre-recorded music, but to most people, it's just a noise backdrop to their daily chores.

They drive Hondas or Fords....and in this economic environment, I'm not so sure that'd be a wise biz move, to try and sell the public on, "... hey, once again, we'll give you higher quality... " marketing goop.... not when they can barely afford to put gas in the tank of that Honda/Ford... and now, food shortage scares...My guess is with these economic swings, we ain't seen nothing yet as to the impact on discretionary spending.

I mean, it's possible that the music/electronics industry could be like other recent industries and say, "Fuck you... we changed the rules... now you have to buy our new IPODs for the 500K, 64 bit sample rate...." and make it stick.  Maybe it's possible when they can connect your IPOD directly to your aural nerves and inject it straight into your brain....I don't think most people would hear the diff on their laptops...not thru 1 watt earbuds.


Edited version of  a Wikipedia page discussion:

As to broadcast engineering in the mid 80's to early 90's, one of the first multiband limiters beyond the Optimod that I recall installing/using was the Texar 4-band.

Some great vids on Youtube of both the Optimod and Texar Audio Prism:


So as you can see, multiband limiting is nothing new.

Recall Orban Optimods did do a sort of multiband limiting but also performed many other FM-related audio processing functions with some versions being the station's stereo pilot generator/processor. We used both in many of the FM's I worked at. This leads me to think that exploration of some of the artifacts (such as intermod distortion) of "misuse" (subjective) of multibands  should be mentioned.

Also, the reason we as broadcast engineers used such multiband devices was to get above road noise, receiver noise/multipath whaps, satisfy the management as to the perception  upon listeners changing stations an apparently louder station, etc... It was also due to the fact that the FCC would monitor stations for peaks over 100% (with no SCA's; 110% max with SCA's) and FINE the station if more than 2 peaks over threshold were detected. Bad stuff. Our local monitoring facility in Langhorne PA was especially strict about going over allocated spectra (typically a deviation of ~ +/-75kHz)

From the FCC 73.1570   Modulation levels: AM, FM, TV and Class A TV aural:

 (2) FM stations. The total modulation must not exceed 100 percent on peaks of frequent reoccurrence referenced to 75 kHz deviation. However, stations providing subsidiary communications services using subcarriers under provisions of 73.319 concurrently with the broadcasting of stereophonic or monophonic programs may increase the peak modulation deviation as follows:

(i) The total peak modulation may be increased 0.5 percent for each 1.0 percent subcarrier injection modulation.

(ii) In no event may the modulation of the carrier exceed 110 percent (82.5 kHz peak deviation).

A great view of the RF spectral content of a typical FM broadcast can be seen here: http://en.wikipedia.org/wiki/File:FM_spectrum_no_IBOC.png
From the subject: http://en.wikipedia.org/wiki/HD_Radio


Also, I did a bit of research (you can see my lab at http://www.ajawamnet.com ) on playback of various audio devices (MP3 player, cheap and not-so-cheap CD players) and what I found causing the crackling with "hot" CD's/MP3's processed to within 0.01DBFS is limitations in the D2A sections and successive analog stages causing the familiar cracking sound. I did this because I noticed that certain playback devices did NOT do this, having sufficient headroom in the analog section.

I've noticed a recent trend in "hot" masters going to absolute values of -0.1 to absolute limits around  -0.3DBFS,  which is below the "crackling" threshold of even the cheapest of playback devices, such as those sold in toy sections of most retailers.

Please note that most of the digital plug-ins and in-line comp/limiters used for digital pre-mastering actually do not "clip". Adjacent 16/24/32 bit words with the same bit train are typically shown as errors in most mastering software; Wavelab allows one to set the threshold in its Global Analysis - Errors tool.

Even Wavelab will show this as a digital "clip" when using their analysis tools (not the real time meters). Therefore most of the limiters I use in the digital domain, such as TC, Voxengo, Waves, etc. produce even the heaviest of limiting with non-repeating maximum word values. One can see this if you zoom into the waveview on any DAW and see that the absolute values are never (or should never be) adjacent. Replicators will usually bounce these "gold discs" upon inspection before going to CD master cutting.

I also believe that this should not become a soapbox for what people think is right or "audiophile" since it has little to do with the factual interpretation of trends towards higher RMS values. I will say that it should be mentioned in very agnostic terms that it is the opinion that ear fatigue may be caused by reduced RMS to Peak ratios, but subjective views of what "sounds good" are just that - subjective.

As to the reasons that this is occurring, a few reasons are mentioned in Katz's book Mastering Audio, one being an increase in ambient noise in listening environments and the resulting effect on what he terms as something like listener threshold.

But my feeling is that we've been seeing a reduction in speaker driver size, and to me, a limitation of the instantaneous ability of a reproduction system to excite a certain volume of air in the listener's environment at all spectra results in the typical listener stating that something sounds better on these smaller, multichannel home systems, small computer speakers, etc... when the program material has higher RMS levels.

The effects are compounded by non-ideal room geometry and dimensions as well as placement of the speakers/drivers by the consumer. It's fairly obvious that these systems exist in non-ideal environments which causes a multitude of problems.

When I worked with the Pittsburgh Symphony, it was interesting to hear the amount of dynamic generated by 108 players at forte during something like Copeland's Rodeo. All of those sound generating elements, able to excite that much air instantaneously was an eye (ear) opener. Same with being onstage at large venues during many of the rock acts I did monitors for. Having that much surface area all working in consort is something not easily achievable in the typical consumer listening environment.

As to an expert, you really should refer to some of the mastering gurus and see if they'd like to contribute. Katz, Ludwig, Smith and others have been doing this for years and have seen the desire of bands, producers, and A&R men evolve to these "loudness wars". In fact, Wavelab now offers a Katz-style meter, which opinions vary, to gauge apparent loudness.

Wamnet 15:16, 8 April 2007 (UTC)


The Original Mastering Engineer article from Wikipedia

This was the original version of the topic I started on Wikipedia titled Mastering Engineer.  By now it's totally unrecognizable due to the constant bickering and posturing of the wikipedia population. 

By now it may even have merged into the Mastering topic.

They all think J-Lo will (INSERT FAVORITE SEX ACT HERE)  if they're the ones that have the final top-dog say... everyone's a rock star.

Oh well. There's so much personal opinion that it's about worthless now. Just like there's good open source software like Blender, there's also a lot of shit that comes with open, GPL'd stuff.

Anyway here it is, the original version:

Mastering engineer

From Wikipedia, the free encyclopedia

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A music mastering engineer is one skilled in the practice of taking audio (typically musical content) that's been previously mixed in either the analog or digital domain as mono, stereo, or multichannel formats and preparing it for use in distribution, whether by physical media such as a CD, or as some method of streaming audio.

Confusion exists as to the term mastering, with the most descriptive of these being pre-mastering to discern it from the actual process of making replication masters for replication/duplication of physical media distributed to consumers such as a vinyl record album or a compact audio disc (CD).

Typical pre-mastering involves processing the audio for the various media targeted. Depending on the medium, it can involve sequencing musical selections, adding necessary timing between selections, adding neccesary coding or information to digital streams for various uses such as PQ codes for CD track indexing, etc.

But most accolades given to mastering engineers is their ability to make a mix consistent with respect to subjective factors based on perception of listeners, regardless of the playback system or environment of the consumer. This can be difficult with respect to the varieties of systems now available that the audio selection may be reproduced on, and the effect it has on the apparent generally-accepted qualitative attributes of the recording. For instance, a recording that sounds great on one speaker/amplifier combination playing CD audio may sound totally different on a computer-based system playing back a low-bitrate MP3.

Add to this the variations in listening environments, (autos, movie soundtrack usage, noisy kitchen, poor acoustics) as well as the placement/coupling to that environment of the sound system in use and one can see the difficulty in providing a consistent listening experience to the consumer.

This is why it's considered as an art as well as an "audio engineering" dicipline.

Another major function in pre-mastering is to ensure that the final product does not exceed the capabilities of the medium used to distribute the final sound recording. Avoiding distortion of various types as well as generally accepted spectral balance and apparent volume adds a layer of complexity to the process.

Currently, trends in pre-mastering has led to higher values of what is, in simple terms, the apparent average sound pressure level or volume of the reproduction when listened to on the varity of consumer audio equipment. One need only to play back a recording made recently, and without changing the reproduction systems gain (ie. volume control) play back an older non-remastered recording and compare the difference in how loud it seems.

This change in what is typically known in the practice of pre-mastering as the use of higher ratios of compression/limiting is probably due to numerous factors, one being the the general consensus in the study of the mechanics of human hearing that the ear is more sensitive to apparent average sound pressure than it is to sound comprising of shorter-duration peak levels. It also appears to be market driven and genre-specific, with popular styles of music tending towards recent trends of higher average program content then say classsical recordings when comparing music distributed in the last 10-15 years.

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