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Is Louder Better?
GoodNicsTken writes "Rip Rowan over at prorec.com did an
analysis of 5 different Rush CD's released from 1984 to 2002. The results show a definite trend in the recording/mastering style from each album. Rip contends that louder is not necessarily better as the record execs believe. The artist however, is often left with little choice in the matter."
Now that is one tough, durable fellow. I would have split my own head open with a .44 slug by the start of the third album.
Air Supply, now there was a real band! ;)
'nuff said! ;-]
"Michael, I did nothing. I did absolutely nothing - and it was everything that I thought it could be."
I'm a fan of the heavy metal genre and I've seen (or heard, more like) many songs that would be absolutely great if they weren't subjected
to the same LOUDER IS BETTER butcher job Rush's Vapor Trails went through. One example is the song "Here Comes the Pain" on Slayer's latest album. I can barely make it past the intro because it simply sounds so terrible. Or if I really want to listen to it, I turn my volume down so my speakers don't peak or bottom out. Turning metal DOWN??? That just ain't right. Damn their sound engineers to hell.
On the other hand, In Flames' latest album entitled Reroute to Remain sounds absolutely beautiful on any speakers I play it on. Same holds true for other Nuclearblast artists such as Old Man's Child and Dimmu Borgir. Kudos to foreign audio engineers!
I always mix to -20 dBFS RMS because louder is NOT better. Headroom is much better.
Hopefully, surround music formats (DVD-Audio & SACD) will convince the tried & true engineers that they don't have to slam recordings at -0.1 dBFS like they've been doing with CDs.
A nice 24 dB of headroom allows for dynamic range in muxic, as well as loud transients. This is something you don't get when your music is an L2 brick.
Jory
Did anyone else shudder at the thought of 5 Rush Limbaugh CDs?
no comment
that the louder speaker system always sounds better. They move a lot of expensive speakers like that.
It's Christmas everyday with BitTorrent.
Well, it's one louder, isn't it? It's not ten. You see, most blokes, you know, will be playing at ten. You're on ten here, all the way up, all the way up, all the way up, you're on ten on your guitar. Where can you go from there? Where?
RRRRR, matey.
Rip Rowan recounts rummaging Rush recordings.
Best Windows Freeware
And don't forget that every Canadian decibel is almost two American decibels.
Another great read here.
My server
More range is better, which can equate to louder "loud"'s, and softer "soft"'s. Just having the record be louder is going to sound like crap on really super-hi-fi systems that can pick up every little thing... you'll hear cats meowing in the studio, etc... I know from experience in the studio!
stuff |
> It's all about the radio. If your song has a lower volume than another one, it'll just sound Lame when it'll start.
> Of course all radios should/would/could normalize their playlists
I just wish they wouldn't blast the commercials out even louder than the music.
Sheesh, evil *and* a jerk. -- Jade
It's been covered in many web publications back in 2002.
Dynamic range problem is real though. This is why you whould avoid mainstream, "radio-ready" artists and bands. Another excellent reason to buy indie music.
If you want to see how bad the problem is, get yourself a copy of the latest Foo Fighters CD and listen to the album with decent headphones. (Grado sr-80/125 or Seinheisers of equal quality). It's just noise.
When I working in a night club, I would receive promotional music on vinyl and cd formats. I could not tell the diff until the volume was way up. Bass sounded amazingly deeper and cleaner from the record. The speakers were flubbering at the same volume from the cd. http://www.howstuffworks.com/question487.htm
Just like how DX-7s and putting huge amounts of reverb on your Linn drum machine were in vogue during the eighties, I think this phase will play itself out. Right now the recording style seems to be centered around, compress everything, auto-tune the vocals, and master it so every track, it feels like the guitars and drums are burrowing into your eardrums. This too may pass. And besides, if people get sick of the excessive mastering trends of today, the record companies can just go back to the master tapes and re-re-master everything, and get everyone to buy all new cds.
No, of course louder isn't better. What rock 'n' roll music clearly needs is more cowbell.
All you need to do is to get the TK421 modification for your amp and everything will sound much better.
my sig
Nope. If you were to try to compress some of those harsh clipped signals, you'd get much better compression than trying to compress a signal with good headroom to it. Go read the article and look at those signals. The peaks and troughs are just way the hell off the scale. When you clip a peak or trough like that, you're essentially throwing away all signal information that was in there. It's really easy to compress something when it's made up of all zero's.
- Give a man a fire and he's warm for a day, but set him on fire and he's warm for the rest of his life.
I think this guy is failing to grasp the implications of the 'loudness' of Vapor Trails. Yes, it is quite 'loud'. It definately SOUNDS louder than previous Rush CDs. But this has nothing to do with the engineering of the album. It has to do with the sound that Rush was trying to make.
Rush was on like a 6-year hiatus. They produced the album (along with another longtime Rush producer guy). Do you think that they would have put out an album that didn't sound like they wanted it to?
Vapor Trails does sound different. There's more distortion, the amplifiers are more overdriven, being pushed to their maximum more... But that is more a style thing than anything else. There's been a lot of Rush stuff that has been very clean, very free of distortion, very clear.
And Geddy Lee, Alex Lifeson, and Niel Peart have said that they chose to make things 'louder' and less clean to give the album a bit more of a 'jam' feeling. They wanted to get back to their roots, and distinguish themselves from the different clean and synthy sounds they had in the '80s.
So... Vapor Trails doesn't sound loud and overdriven because it is engineered poorly, or because not enough effort went into producing it... it sounds that way because that's the sound Rush was going for
And for the (slashdot) record, Vapor Trails has generally been recieved well by fans, and has gotten very good reviews. And I like it, so you KNOW it's good stuff.
no thanks
Radio does have automatic limiters. Listen to a rock station sometime, it all comes out about the same level, despite the different levellings of the individual recordings. This was in the article, btw.
-Looking for a job as a materials chemist or multivariat
Here's a link (reassemble as needed).
n g. html
http://www.personal.uni-jena.de/~pfk/mpp/clippi
First of all, the article is dated September 2002, though that doesn't make the writer's concerns any less valid.
Second, this has been going on for almost twenty years, starting around the time digital tape decks (like Mitsubishi, Sony, 3M) gained wider currency in recording studios. Digital audio sounds really harsh when you push recording levels, as opposed to analog tape, which has a "softer" limit.
Rowan makes a very valid point: radio stations are notorious for compressing their feed, mostly to get the hottest signal within their transmitter's power limit. Television stations are even worse. I recall taking a road trip with my band in a rented van that didn't have a cassette player; we were at the mercy of every Top-40 station and all of them were playing Phil Collins's "Sussudio" every ten minutes. Some of the stations flattened the signal so much that we thought it was some sort of remix just for robots (the drum machine was at least twice as loud as the lead vocals).
Where I don't concur is Rowan's placing the blame for this on the labels. True, the A&R people are the ones who have right-of-refusal on the final mix, but you can't let engineers, producers, and the mastering lab off the hook. I've been on the other side of the glass and I know that I've been guilty of patching compressors into a channel to keep the kick drum at a managable level, make up for a singer's lax microphone discipline, or "punch up" the final mix. Note that I'm not blaming the musicians; they do whatever they have to in order to get the track on tape. If that means Joe Frontman is going to sway back and forth like Bill Gates at a deposition, so be it. It was my job to deal.
Finally, not to sound too much like a Luddite, but back in the analog days, there was a limit on the number of effects you could employ, the limit being the number of physical units present in your studio rack. Now, with ProTools or Cakewalk, your limits are RAM and CPU cycles, both of which are cheaper to expand than buying more compressors, limiters, gates, reverbs, etc.
k.
"In spite of everything, I still believe that people are really good at heart." - Anne Frank
Indeed. There is a point after which you begin to clip the music and reduce its dynamic range. If you record the damn thing too high, I will never be able to play it loud without distortion.
My brother tought me this 20 years ago when he showed me how to make tapes. I would sit there and stare at the VU meter throughout the WHOLE song, turning down the record volume slightly every time it hit red. Then rewind the song, and now with the volume properly set, record it.
Later I learned to let a bit of red slip in there, to taste. If its loud and distorted, its just pure garbage.
Personally I do not like rock and roll. But if its lound and 'clear' I can dislike it with a sort of appreciation...
I tend to have a theory that perhaps the Music Buying Public is starting to get tired of all these empty, manufactured pop bands that come out of Disney. That and a lot of the mainstream stuff that was based on the Seven Formulas To A Perfect High-Selling Pop Song (or whatever that was) (read: what 80% of the population buys, because 80% of the population buys it) has just become way too tired after 25 years. I think the music industry's own marketing is thier biggest problem.
/me gets off his soapbox and offers everyone else a try
That said, let me step on my soapbox for a sec..
As a music buff, a musician, and someone who's seen the musician's side of the music industry in nearly all its forms (garage, stage, touring, studio, etc)...
I will first say that getting music recorded is a fairly long-winded and convoluted process...
1) The sound you get out of the instrument's amp in the studio is not what you'll get on tape
2) In the mixing process, there is a great deal of EQ'ing, Compressing (this is what gives the LOUD), and various other things to get things to come together in a certain fashion. When all is said and done, the sound you had on tape before is now going to be totally different.
There are many many schools of thought on how best to master a recording. Some go for atmosphere, some go for candid honesty, some go for a super-polished sound, well, you get the picture.
However, the trend i'm seeing lately with a lot of old albums, is that they're getting remastered in a modern studio with the attempt at "Updating" them. I don't know if this is something rookies cut thier teeth on or something, but i've got a lot of horribly done CDs. I do realise that the difference of listening to stuff on my old, worn out vinyl or tapes as opposed to a CD will be fundamentally different just because of the analogue/digital conversion.
Sabbath albums that are gated so hard, that everything is muffled to hell, but the vocals are enough to spring THE ENTIRE MIX open and everything distorts.
Maiden albums where someone took the effort to attenuate the feedback from the guitars. This really blew me away. like "Dewd, Adrian Murray WANTED that there!"
I've got a few hendrix and yardbirds albums where everything was squashed into oblivion with a compressor/limiter (failed attempt at making something LOUD). Yes, the album is loud, but it doesn't *breathe*.
I've got a fleetwood mac album where everything sounds cold, thin and empty. Too much noise reduction. Noise reduction being my biggest beef.
IMHO, the bass guitar rattling the snare drum in an intro, the 60hz hum of the PA, all the delicious lil freaks of sound that come out of guitar amps..... to me, that's just as much a part of the music itself. I love the *noise*. My old vinyl was full of it.
When stuff gets too `polished' i think it loses too much of it's `soul' and becomes a little too mechanical. I don't expect everyone to agree with me on this, though, so to each thier own.
Starting to get spooky, isn't it.
As copyright owner of this comment, I authorize everyone to defeat any technological measure which limits access to it.
I'm a fan of the heavy metal genre and I've seen (or heard, more like) many songs that would be absolutely great if they weren't subjected
to the same LOUDER IS BETTER butcher job Rush's Vapor Trails went through.
The article mentions that artists usually don't have a choice in the matter, but Geddy Lee himself did Vapor Trails. He stated in interviews that he was having breakdowns because everything was digitally clipping, but that he was reassured that it sounded okay by the rest of the band.
"Sufferin' succotash."
I've always wondered about why (more or less) permanent audio storage formats like CDs or DAT use linear PCM when it's fairly clear that the human auditory system uses a logarithmic transfer function. Wouldn't we be better off using 16 bit logarithmic samples instead of linear samples on CDs and such?
Note also that the article points out the legitimate uses for pushing up the volume without any distortion. For example, many pre 1980s recordings are now getting a second workover: the original release was on vinyl, then there was the simple 1980s digital transfer to CD, and now many classical recordings (e.g. most of Rudy Van Gelder's recordings for Blue Note) are released a third time after 24 bit remastering and mixing. (Plus there are the Japanese 20 bit releases from the 1990s.) This does make sense, since you when transfering your final 24 bit mix to a clunky old 16 bit audio CD, you need to make sure that you keep the volum as high as possible without introducing distortion, coz if you don't, you lose detail in the softer passages due to the fact that you have to drop the least significant byte of each sample. So louder is in fact better, as long as you don't clip the peaks.
Marklar: marklar
And I quote "No computers were used during the writing, recording, mixing, or mastering of this record"
"All songs on this record recorded to eight track reel to reel at Toe-Rag Studios, Hackney, London, England by gentleman Liam Watson in Apil 2002 except track 4 recorded at the BBC Maida Vale studio by Miti"
I haven't seen liner notes like this (i.e. referring to the recording process) on a rock album in a really long time.
This was the same album that was sent to radio stations in vinyl only, the speculation being, they were trying to avoid it being uploaded to a P2P network. But accoring to an interview, vinyl is their preferred listening medium, and they wanted people to hear it in that same manner.
I have both versions of this album, and I must say, that the vinyl disc, on a VPI Aries Scout and a tube phono preamp are not subtle.
And the detail! It sounds glorious!
You hit it right on the head. The trend in radio lately has been to compress the hell out of the music they broadcast, and in turn, record companies have jumped on the bandwagon with CDs. Most music consumers think louder sounds better, and so that's what sells. It kind of makes sense even -- just listen to a recent mega-compressed track at a comfortable volume, then listen to a track from an old CD at the same volume. The older one sounds weaker, but only because it is softer. Adjust the volume again and it probably actually sounds better. But most consumers don't care enough to make that realization.
Back in the early 90s, a remastered CD was something that actually sounded much better than the initial digital transfer of a classic album. Nowadays, remasters accomplish two things: compressing the music until it's all one uniform LOUD volume, and lining the pockets of the record industry as die hard fans buy the same albums again.
Of course, this trend is not all bad. Not hearing soft sections of music in the car is a legitimate problem. I won't listen to classical music in the car because of this - I tend to stay within the rock genre because of this and only listen to classical and jazz in the quiet of home. It's too bad that record companies are now "solving" the problem by giving us this "one volume fits all" compression now. The ideal solution might be for car stereos to start including some sort of compression circuitry so that you can hear more of a tune over the road noise, but you get to hear it in its full dynamic glory at home. Heck, other things like TVs and DVD players could use this too. Sometimes a TV show or DVD will need some compression so I can hear the quiet parts but don't piss off the neighbors during the loud parts! Either that or maybe some sort of new audio format with two versions of each audio stream - normal and compressed. Of course we already have SACD and DVD Audio, yet another new format is just what we need...
Say hello to zMac.
I work with ADCs and DACs all day. Your first pass answer of 96dB is correct for DC characteristics. However, sinewaves introduce some differences.This ADC is a darn good performer. You will notice that the SFDR (Spurious Free Dynamic Range) is -101dB, while the THD is -99dB. Also, its Signal to Noise is -92dB, while the theoretical best is -98.08dB.
In fact, a small amount of noise actual can improve the signal representation! But that is a rather long discussion.
No, I don't trust in god. He'll have to pay up front, like everybody else.
Ok, here's what happens.
First off, while everyone bashes analog, the analog signal is what you want to measure. When you convert to digital, two things happen:
1) sampling in time. The sampling in time reduces the maximum frequency that can be represented to half of the sample rate. This is not a big deal, since you really can't hear much over 22KHz (for CDs) anyway. Just make sure that you have a good lowpass filter so that signals don't alias.
2) quantization. While the analog signal has an infinite range, you would need an infinite amount of bits to represent each signal as digital. While modern hard drives have gotten enormous, they still are not infinite. So, quantization restricts the valid levels to a finite number, and also restricts the minimum and maximum levels that the digital signal can represent.
Generally, for signals with a large amount of frequency content (what you kids call 'music' these days), there is a large amount of peaks. However, the peak is not what gives the impression of loudness. The effective amount of power, referred to as the RMS, is a better indicator of loudness than the peak levels. The peak of a sinewave is 141% of the RMS of a sinewave. More complicated signals will have a peak-to-rms ratio much higher (1000% or more).
So, when you are digitizing a signal, if you keep the input range of the converter constant and keep increasing the input signal amplitude, you will be increasing both the peaks and also the RMS levels. Once the peaks hit the maximum level that the ADC can represent, the peaks start getting clipped - but you can still increase the RMS. However, as you start clipping the signal more and more, you increase the amount of distortion in the signal.
No, I don't trust in god. He'll have to pay up front, like everybody else.
I've been listening to the VAPOR TRAILS CD in the car, and I thought I was hearing clipping. Knowing that Rush albums are among the most meticulously crafted in the business, it never occured to me that the CD might have been mastered clipped, but that is exactly what seems to have happened.
wow, your reply was factually pretty off-base. let me school you on a few points, guy:
1) please for the love of god, SPELL THE WORD "NYQUIST" CORRECTLY.
2) latency is dependent on how large the input buffer is, and how often it is emptied. doubling the sample rate should, therefore, have the same effect as halving the buffer size, as it forces data to be clocked out of the buffer twice as fast. this is why the latency halves when the sample rate doubles.
3) let me quote you here.. "What does 96Khz do for you as a consumer? Well, with 44Khz audio interfaces, consumer level stuff is not always the most perfectly matched. As such, aliasing happens. Get 2% phase in a cap or resister and then multiple this over the signal path...Nyquest? Nyquest ain't gonna help you out when the alising is dipping into the audible portions of the sound. Remember 44khz gives you 22 IF EVERYTHING ELSE IS PERFECT." --- that was a collection of pseudo-babble and technical non-sense. total shit. sorry, guy, but i had to call you on that. aliasing is a phenomenon wherein the input signal is sampled at a rate unsuitable with the nyquist cutoff- in that instance the sample is "faked", and the sampled signal that results is of a lower frequency (but usually harmonically related to) the source sound - it is a mis-representation of the input signal, like jaggies in a computer generated picture. it has absolutely nothing to do with phase smearing, which is more of a time-based issue between two or more distinct signals canceling each other out at an audible rate.
4) I happen to own a Kurzweil K2vX (basically a K2000 with the sampler module, orchestral and contemporary ROM blocks built in). once again, you've got a logic flaw- in actuality, the kurzweil samples MANY of it's instruments at MUCGH LOWER RATES than 48k. SERIOUSLY. just click the "MASTER" soft button, then SAMPLE, then check out some of the built in samples-- the piano is sampled at about 22k. this is because the most relevant portions of the instrument in question need only to be sampled precisely. your kurzweil sounds good because of the relatively clean internal processing and D/A converters that are built into the keyboard (MUUUUUUUUCH better than the coverters built into other keyboards of the same ilk).
5) in theory, yes, 96k was made to correct issues with regard to aliasing. however, the cost of incorporating this high frequency sampling rate is a high one-- companies whose converters are cheaper will be unable to accurately pace themselves during the sample process (perhaps a cheap crystal) and will produce much more jitter-- each sample of a 96k wave is supposed to be evenly spaced.. like a drummer, it has to "keep the beat". well, imagine if the drummer was epileptic. that's what happens with cheap converters). so the introduction of 96k products in the consumer field is a nightmare for the time being.
6) with regard to dual inline converters that sample at 44.1, but "double up"-- if done correctly (meaning, if CLOCKED correctly.. see above for the jitter issue), than there is NO DISCERNABLE DIFFERENCE in the QUALITY of the sample-- if the converters (2 of them, remember?) are assigned as A and B, and each fires at the correct interval (A B A B A B A B), than they can "interleave" a 96k signal with NO PROBLEMS and NO DEGRADATION OF THE SIGNAL ITSELF. i have no clue where you got that idea.
7) you made a bit of a joke regarding word clock.. but i think you may really wish to research the definition, because perhaps buying one of those would improve the sound of your recordings FAR MORE than dicking about with "nyquest" theorums and fantasies.
buh bye.
First, minor point - 96 kHz sample rate gives you 48 kHz theoretical bandwidth - Nyquist frequency is exactly one-half the sample rate. Not 44 kHz.
So, here's the real point. Higher sample rates allow you to pass higher bandwidths through the ADC (and theoretically through the DAC). However, those higher bandwidths get shrunk when they hit the amplifiers (consumer, even pro-sumer products rarely use high-bandwidth amplifiers), further shrunk when they hit your speakers (know how many speakers will produce anything above 22 kHz? Simple answer - not yours. Unless you spent several tens of thousands for your high-efficiency ribbon tweeters), and even further shrunk when they hit your ears (though, some people can indeed hear above 20 kHz. I can hear up to 26 kHz, but then, I've never been to a concert or club without earplugs). The acoustic coupling they claim happens in mid-air (which is true - put a 40 kHz tone and a 44 kHz tone out - you get difference tones at 4 kHz, 8 kHz, 12 kHz, etc.) only occur if your speakers can get those high-frequency tones out in the first place... which they probably can't.
So, what's the real point?
Better anti-aliasing filters on the ADC side. If you are sampling at 44.1 kHz, under the Redbook standard, you have to be down 40 dB at 22.05 kHz. However, you want to pass 20 kHz with no filtering, which means your filter has to be as brickwall as possible (about -200 dB/octave... sheesh!). 3rd order filters can't even do that properly, so most anti-aliasing filters start rolling off around 16 kHz, some even earlier (especially in digital video cameras. I know, I've tested 'em).
So instead, set your filters to be down 40 dB at 48 kHz ('cause you're sampling at 92 kHz). Now your filter only has to be about 36 dB/octave to pass 20 kHz untouched, and that both increases your flat bandwidth and decreases phase distortion (the -3 dB point is a 45 degree phase inversion, and every 3 dB after that is another 45 degrees of delay).
That's why sampling higher improves things - no brickwall filters.
-T
Sorry, but you seem to misunderstand the situation. What the sampling theorem tells us is that as long as the input signal is bandlimited to frequencies below one half the sampling frequency, it can be reproduced exactly by the DAC. (nb: this is for samples which have not been quantized) The reconstruction is taken care of by what is commonly known as a reconstruction filter. You are correct that the samples of a sine wave near the Nyquist frequency will look like a triangle wave, but once passed through the reconstruction filter what comes out is the original sine wave.
Note that the requirement that the signal be bandlimited means, for example, that one can not have as input a triangle wave near the nyquist frequency, because, as you correctly stated, a triangle wave of such a frequency contains harmonics which are greater than the nyquist frequency. Typicaly, they would be removed by an antialiasing filter at the input to the ADC.
Of course, I realize that it is not possible to implement an ideal reconstruction filter, and also that quantizing the signal introduces distortion which it is not possible to remove.
Many different people do 'Mastering',. and each one does it in his or her own special way that they want to convince you is better than everyone else's. The main steps in mastering are eq, compression, and level matching. Very few cd's are printed with clipped samples, because this data is out of range, it can't be reproduced, if the playback of the cd results in a clipped waveform it could be either a perfectly recorded clip or it is a failure of your hardware to faithfully repoduce the waveform as it is encoded on the disc. Most generally it is the latter.
When finished tracks are sent to be 'Mastered', they are usually compressed a little bit, or a lot, depending on the taste of the Mastering Engineer. Compression in this case doesn't refer to encoding audio in a compressed format, rather a compressor is a dynamics processor, with it you can set a threshold above which the sound will be modified based on a ratio like 2 or 3 to 1. So for a 2 to 1 ratio any sound that is above the threshold will be reduced by half.
This was initially done back in the old days when you had at best 45 dB of dynamic range to work with on your recording medium, a very noticable noise floor, and material with a dynamic range of 120 dB (Live Rock). Obviously you can't stuff 120 dB into a 45 dB (cassette tape(if you are lucky)) dynamic range, So the material was compressed to fit within the dynamic range. Also because of the quality(lack thereof) of consumer audio equipment and the previously mentioned very noticable noise floor, most music is compressed into the top 3-5 dB of whatever medium it is recorded on.
Nowadays, we have a playback medium with a 96 dB dynamic range and close to a 96 dB noise floor, but because people got used to the way it used to sound, they want to keep hearing it that way. Pretty much the only recorded materials that truly benefited from the increase in dynamic range allowed by CD's and digital recording are orchestral works, and the people that listen to these avidly, and care about the recording truly reflecting the performance, still want more!
The other aspects of 'Mastering' are a great deal more subtle, equalization and level matching between tracks are things that most people do not notice unless it is done badly. At the end they turn the result up to the top of the mediums allowable dynamic range and start printing tens of thousands of them at a few cents apiece.
If you think a cd has clipped samples recorded on it the best way to check is to rip the track off the disc into a PCM (Non Lossy, Non compressed, Non MP3)format at 16 bit/44.1 (Redbook native format) and look at the samples in question with a wav editor. If you have blown up the waveform to the point where you can see a single sample, and the tops of the waveform are at the cieling and flat, then complain to the recording engineer, because it is probably his fault.
BTW make sure it is a clean non scratched cd, any unrecoverable data loss can appear as a clipped waveform, and is heard as such depending on the smoothing filter on the output side of your cd player.