Does Going Digital Mean Missing Music?
arlanTLDR writes "The Seattle PI is running a story about how the MP3 format is the sign of a musical apocalypse. Apparently, many top music producers are 'howling' over the fact that files in a compressed format contain 'less than 10 percent of the original music on the CDs.' Is this just sensationalist FUD, or is there something to the assertion that listening to an MP3 is like hearing music 'through a screen door?'" The article mentions that the iPod and its cheap earbuds bear some of the responsibility for rendering this degradation in sound quality less objectionable.
Bad mixing. I can't find the link right now, but many people have complained about how CDs are being produced by mixing things loud and the sound getting clipped. Add to that most consumer CD players completely process the CD signal to hell and gone then they play it through cheap-ass head phones so seriously, the consumer has already lost a lot of quality. Most listeners won't notice the difference because of their playback set-up.
Of course, some people are now going for the "super bitrate" MP3s ripped directly from CDs, but they are the rare ones.
Also, if the mass market really wanted higher audio quality, don't you think any of the CD successors would have taken off already?
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most people don't care about the sound difference between mp3 and cd. hell i have friends who like the over compression of FM radio. i can tell you the difference between 320kbps mp3 and a cd, and anyone who has a quality sound system can hear the difference as well. solution: audio reconstruction. there are many algorithms out there that can simulate the missing highs and lows which is satisfying enough for most people (i have a friend who can't stand the way mp3's fuck up guitars and high hats).
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ipods have a few million users as a base, i bet at least 25% (probably way more) use the $0.50 earbuds that came with them. they suck, yet the users are fine with it. apple keeps selling ipods with shitty earphones, users accept the way music sounds. hell even dell's $20 speakerset with subs get rave reviews from my friends who live in college dorms......until they hear my $1300 5.1 setup
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Quite right. Maximum PC had an article about a year ago where they pitted 4 people (A teeny-bopper or two and at least 1 audiophile) in a "Guess the Source" contest. They had a selection of songs and played 4 versions of each ranging from 160kpbs mp3 up to flac and uncompress wave on various sound systems (iPod earbuds, expensive head phones, expensive stereo system, etc).
As I recall, nobody could really tell the more often than chance would predict. The audiophile did slightly better, but nothing to shake a stick at.
I don't know about guitars (I've never heard one compressed that wasn't too buried in the mix to identify), but I can't stand the way AAC screws up cymbals. Anything that naturally has no tonality tends to be massacred by lossy compression. If you can't hear the difference, you probably can't hear high frequencies.... It is annoying to me even with cheap earbuds in a quiet room. In noisy environments, I can't hear the difference, of course, thanks to the masking effect of everything else. Whether you can hear the effect or not depends largely on whether you are actively or passively listening to the music.
Honestly, I don't mind the earbuds. The proximity to the ear makes up for most of the low frequency loss associated with a small diaphragm, so they sound acceptable. They aren't God's gift to man or anything, but they aren't nearly as awful as you make them out to be. Now computer speakers... those tend to be universally abhorrent. No bass response whatsoever, so they sound like tin can telephones.
As for the $20 speakers with subwoofers, they get rave reviews mainly because most people have never heard good speakers. Compared to a set of 6 inch drivers, yeah, they probably sound great. You actually have deep bass response. Compared to a pair of properly tuned 3-ways with 12 inch drivers, they sound like ass because you have probably a couple of octaves of upper bass to lower mids that are mostly missing because it's too high for the sub to generate it and too low for the tiny 4 inch (or smaller) main speakers to generate it. Compared with my studio monitors, they're laughable. The problem is that most people have never heard speakers with drivers over about six inches... maybe eight. Oh, yeah, and most people don't have any hearing above 14 kHz anyway, so those tinny little speakers sound good to them. :-D
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Here is a newer test or a rehash of the older test.t _rates_pay_off
8 _bit
http://www.maximumpc.com/article/do_higher_mp3_bi
This test showed it was hard to pick the differences, but they conclude using vbr with a higher bit rate would improve the sound quality.
Here is a comparison of earbuds using apple's aac formats at 128 vs 256
http://www.maximumpc.com/article/itunes_256_vs_12
Cheap ear buds expose the differences in compression levels, while expensive earbuds make it hard to tell the difference.
Good info. Also agree, lows and highs are what suffers with poor compression. Even with my awesome-for-the-price $20 sony headphones I can really tell a difference between mp3's ripped to 128 vs 320 vbr. What really stood out to me was the lows sounded so much more alive.
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I'm not at all convinced. I have an awful lot of music encoded at a mere 160kbps, and I can't usually tell which I'm listening to. Of course, I don't have an astoundingly great stereo... But since I can't afford one, what do I care? In the world I actually live in, no, I can't tell the difference; it's swamped by other noises.
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The JS encoding usually used by MP3 encoders isn't doing what you seem to think it does. It's simply a more efficient way of encoding exactly the same thing - by, rather than encoding left and right channels, encoding a sum and difference [usually known as M and S for Mid and Side] of the left and right channels. This actually HELPS the sound quality sometimes, by reducing phase-shifting in the mono part [i.e. identical in both channels] of a 2-channel stereo signal.
With a good enough algorithm most people, including those with well-trained ears, will not be able to consciously distinguish the two sounds. But that does not mean that these people don't subconsciously react differently to them. One way to measure that might be to measure brain activity in various regions of the brain, which is exactly what this article mentions. The problem is that that type of test is always going to show a different reaction which is something the makers and users of audio codecs often don't want to hear.
The major problem here is what does the brain activity data mean? Even if you can see a difference in brain activity for a 16 bit/44 kHz PCM file verses a 128 kbit/sec VBR AAC file how do you determine if one format is preferred over the other?
You end up still falling back on subjective measures, it's much simpler to have a large number of participants and then ask them questions like, "Which recording did you prefer?" The data from a properly run survey is much more likely to yield meaningful conclusions than scans of brain activity. We are, after all, dealing with music - a highly subjective art form.
One notable feature of DSD is that dynamic compression occurs at higher frequencies yet the frequencies are able to be reproduced accurately. Contrast this with PCM where the dynamic range is fixed (i.e. 16-bit, 20-bit, 24-bit) but at higher frequencies the tonality is not as pure because it's impossible to represent anything other than a square wave at the nyquist frequency which is exactly 1/2 the sampling rate. Of course, a filter is applied to make that into a more pleasant sine wave. Now consider a frequency that is not exactly 22.05 kHz but perhaps a little shy of that. It's almost impossible to represent this accurately with PCM. The result is that you actually get a slightly oscillating frequency somewhere around the original frequency.
What you are describing is a phenomenon known as aliasing.
I'm not sure you completely understand how the Nyquist-Shannon Sampling Theorem works. It boils down to the fact that as long as you sample at a rate greater than double the maximum frequency you want to capture, you will get no aliasing. This means that if you sample at 44.1 kHz then all frequencies below 22.05 kHz will be represented accurately. If you sample a frequency just shy of 22.05 kHz you will NOT "get a slightly oscillating frequency somewhere around the original frequency".
It is true that DSD has a variable dynamic response that depends on frequency but that works both for and against DSD since higher frequencies tend to less accurately represented than lower frequencies. In fact there is a lot of discussions (PDF file, see page 8, section 3[c]) that conclude that the current implementations of DSD produce worse quality per bit than an equivalent bit-rate PCM sampling. There are solutions to these problems but they are very complex and involve a mix of DSD and PCM sampling methods, so much so that the line between DSD and PCM blurs considerably.
This has a serious effect on how an album is mastered. When the target format is CD the producer can cause the CD player to output extremely loud high frequency sounds though not particularly accurate frequencies. This is reflected in the current crop of music which is often extremely loud and to many ears just sounds like a bunch of noise. Metallica's self-titled black album was one of the first to use severe dynamic compression to make the album sound super loud. Comparing it with modern CDs we can see that that album was relatively tame.
Again you are mixing up sound levels with frequencies. Severe dynamic compression basically limits the number of sound levels which are utilized,
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I've used laptops to create music. Direct synthesis to CD, live tracks via an ACD, mixed using headphones directly connected to the little skanky computer audio-out. Even then, with a decent set of headphones, you can readily distinguish an AIFF stream from an MP3 in a duoble-blind test. A studio-quality signal chain sounds cleaner for this kind of work, but it's a couple of orders of magnitude more costly too. I did early mixes on the computer and mastered in a studio (good to have family members in the business).
So you don't even need the $500 stereo to tell the difference.
Along with noiselike sound sources such as cymbals, lossy compression also does a number on sharp transients. My own pet peeve is what happens to pick noise on acoustic instruments, as well as the "swoosh" effect on higher-frequency percussion events. Even at 256kbps you can hear mushiness. And my high-end hearing is not what it used to be-- I don't think I can hear much above 18khz anymore.
What I wonder is how many engineers are now recording and mixing so that the song will sound OK even when it's mashed into a 128kbps MP3. Similar to how they used to listen to trial mixes on shitty speakers from AM radios since that's how the kids would hear it back in the day. You think there was an esthetic reason for all that compression? It was making the best of the limitations of the medium.
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You gotta think about just what they're putting that music into, highest quality in the world won't matter for shit if your putting it into overdriven dime store earbuds
Compression meh, for some things you can tell the difference, IF you know what your listening to and know your equipment
Yes, an MP3 is probably about 10% the size of an uncompressed file, but MORE than 10% of the INFORMATION is there, not all of it is there, MP3 is a lossy compression scheme, yeh you lose data, BUT there are lossless compression schemes, and they still give you a file size smaller than uncompressed data.
Does any of this MATTER? um nope not to me, whoop de do the industry complains, does i care? NO. Should the rest of the world care? Well if you are an audiophile you probably already knew about it and already listen in a way that works for you. If your not an audiophile, probably doesn't matter much, your music sounded fine yesterday, should sound fine today.
NOTE:
i am a bit of an audiophile, good etymotic earphones, high quality cartridge in the record player, good cartridge amp and low noise preamp (with hand picked parts)
ALL of the music on my ipod is compressed, jethro tull sounds great, so does blue man group, Manhattan Transfer, and panic at the disco
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That's not really what I'm aiming at. But I largely agree.
Chances are that your old classical music LPs have been compressed for dynamic range too. But that was simply due to the fact that an LP simply doesn't have the dynamic range required to fully record classical music. So the compression was a technical matter. Compare it to the need of tonemapping to display a perfect HDR image of a physical scene on a monitor or print.
Today however, pop music is compressed for dynamic by design, simply to make it sound louder than other songs when played at the radio or from a jukebox where the volume knob is left untouched. Compare it multiplying your image by a factor ten simply to show it "brighter" than on the monitor next it. It will "appear" brighter if you don't look too closely, but it will have lost much detail simply because it clips to your monitor brightness.
Modern pop music may not have the same dynamical range as classical music, but it still has more potential than you find on most of today's CDs:
(a) live performances _do_ have the punchy bass drums and highlighted snares lacking on the album, because the PA can deliver what's requested without clipping. Unless your band is called Cradle of Filth (they suffer the EVERYTHING LOUDER THAN EVERYTHING ELSE syndrome even on stage).
(b) pop albums of 20 years ago don't suffer this problem. They still have lots of headroom. Even trash metal records (Metallica - Kill 'Em All) behave very well with lots of dynamic range and lots of headroom. Yet, the moment they are remastered, that's totally gone. There's suddenly clipping all over the place. It's still the same music!
So it's not the music. It's the mastering.