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Speculation On a Lossless iTunes Store
DrJenny writes "C|net UK has up an interesting blog post predicting that within 12 months Apple's iTunes Store will include a download center for lossless audio. This would be a massively positive move for people who spend thousands of dollars on hi-fi gear, but refuse to give money to stores that only offer compressed music — they could finally take advantage of legal digital downloads. The article goes into details on how Apple's home-grown ALAC lossless encoding relates to FLAC, DRM, and the iPod ecosystem."
Speculation On a Lossless iTunes Store
Lossless? I thought the iTunes store was a loss leader?
The theory of relativity doesn't work right in Arkansas.
Forget "lossless" when you've already lost so much of the original wave by mixing it down to 16-bit 44khz stereo in the first place. I'd rather have something that started out with a higher sampling rate/etc, but with good lossy compression to pull it down to something that doesn't require DVD-type storage for a single album.
I have seen the future, and it is inconvenient.
Hope this happens. After transcoding my CD collection to FLAC to arhive it, I now regularly batch re-encode to smaller and smaller bit rates using new releases of lossy encoders. AAC has gotten much better (esp AAC-HE) over the years to the point for a portable player, 48kbs is perfectly acceptable to my ears. With a 16GB iPod Touch, I could see buying music from the iTMS in some lossless format and transcoding to get my entire collection all on a small, flash memory player.
From the blog:
"And now I have an inkling Apple will add lossless music downloads to the iTunes Store within the next 12 months."
Translation:
I have no fricken clue that this will ever happen, but because I think it'd be cool if it did, I'll go ahead and blog about it.
If someone says he and his monkey have nothing to hide, they almost certainly do.
import system.cool.Sig;
My kingdom for a mod point.
Most CDs have about 10-19 songs and range in price from $10-$15 (at least the mainstream ones). That works out to usually $0.99 a song. The last album I bought was Timbaland: Shock Value. 17 good songs for $12.
Isn't it amazing that 25 years after the release of the CD, we're excited to finally have a way to buy DRM free, lossless, digital music? If this happens, we'll be back inline with 1982 technology.
Sorry, Nyquist's theorem states that you can accurately represent frequencies up to 1/2 the sampling rate. Assuming you are a human and not a dog, you can not hear frequencies above 22khz. As for 16 bit, nobody uses all that dynamic range anyway. So 16bit/44.1khz is entirely good enough for listening.
Now 24/96 has its uses if you're mastering something, so that any errors introduced in the mixing process are below the quantization error in the final 16/44.1 product.
Give me Classic Slashdot or give me death!
Forget Apple... I updated my iPod's firmware to Rockbox (which natively offers several lossless formats, and a slew of other features) and haven't looked back.
I did this for 3 reasons... 1) iTunes stopped supporting Windows 2000. (Yes, I know it's old, but I don't have to deal with the stupid BS Microsoft has built into XP, like WGA). 2) The 1.2.1 Apple firmware for iPod Videos gave me trouble with a bunch of my MP3s--cutting off the song at the 75% marker and refusing to seek within the track. (Of course, the catch-22 is that I can't get a newer iPod firmware from Apple since they refuse to support W2K). 3) I never liked the way iTunes worked in the first place...
I don't hold out much hope that a lossless format sold thru iTunes will truly be lossless. After all, converting an LP to 16-bit 44.1KHz WAV is, by definition, lossy (but outside of the perceptions of 95+% of the people out there)... To add, part of the reason that iTunes even sells DRM-free music is because the record companies can say "if you want higher quality, buy the CD or, better yet, vinyl!" So, I doubt many record companies will be selling uber-high-quality lossless tracks through iTunes...
Windows 3.1x calc: 3.11 - 3.10 = 0.00
So they lock down these files with DRM. Then DVD-Jon (or someone else) comes up with a DRM-stripping program for the files.
Then people can re-encode the files to their format of choice. But by then, most consumers have said "fuck it" and decided to just download their format of choice directly from p2p or usenet because it's easier and simpler than paying Apple and still violating the DMCA just so the music they paid for will work on the audio player they own.
Oh wait, that's already the status quo... Never mind.
24 bits per sample, cool. With you all the way.
But, 96 KHz sampling? You do know the Nyquist theorem, don't you? You are aware that top human frequency tops off around 20 KHz, right? That 48 KHz, even with 24-bit precision, should take care of all sounds possible for the human to hear?
I've had audiophiles* just snub their noses at mathematical proof and regrettably inform me that I do not have "the golden ear." I wonder if there have ever been any research on whether self proclaimed audiophiles REALLY have magical hearing.
(* You didn't say you were, don't take it personally. When I see super-high sampling rates bandied about I get a little red.)
More Twoson than Cupertino
...make some noise; here's one place to start: http://flac.sourceforge.net/itunes.html
almost everyone else distributing lossless (except musicgiants) is using FLAC and/or WAV. it's supported by almost all s/w except itunes, hell you can even get wmp to play FLAC with some work.
re:TFA, lossless is not directly about quality, mp3 and aac both can be perceptually transparent for the most part, it's about (depending on your personality) perceived quality or format independence -- i.e. being able to transcode to the format you need without quality loss.
FLAC - Free Lossless Audio Codec
Sure I might buy something in Apple's lossless format from iTunes, but
A - If I'm going to pay extra for DRM'd lossless, I better get the cheap lossy version for free (for my phone, wife's iPod, whatever) because paying them to compress a song for me is ridiculous,and
B - It will be a moot point if the player won't play all the FLAC I already have, because I won't own the player. It's why I don't own one now.
Operator, give me the number for 911!
Enough with the 24/96 wet dreams. Yes, 24/96 does offer real advantages for mixing houses in terms of being able to normalize levels generated by different sources and reducing the complexity of filters. But 16/44.4 is perfectly fine for home audio playback.
What does >16 bits get you? More dynamic range. BFD. 16 bits gets you (realistically) 90+ dB of dynamic range. Unless your listening room has a background noise level of 20 dB or less (trust me, it doesn't), you're not even enjoying the true benefit of the 16-bits you have now.
What does > 44.1kHz sampling give you? Wider frequency response. BFD. Let's assume that most people have good hearing beyond 20 kHz (very few do). Let's assume that most music/movie content has lots of information above 20 kHz (some do, most don't). Let's assume that your speakers can reproduce signals above 20 kHz (some can, most can't). There is still the issue of how you get that > 20kHz info on your recording on the first place. You see, most microphones don't record signals out that high, and of those that do, they only do so over a very narrow angle. When we have tech that can produce mics that are omni-directional above 20 kHz for reasonable costs then maybe you'll have an argument.
Let's deal with the loudness wars before we start worrying about 24/96.
Am I out of the loop? I was under the impression that most piracy was of the low quality mp3s that suck on any high end audio gear.
Lossless is a great idea and may open up a new market to the iTMS, but I can't image it's going to offset piracy. I'd think it will offset physical CD sales.
Help! I'm a slashdot refugee.
the article claims that apple won't go with FLAC because we're against DRM. I don't think so; if we're to believe Steve then he's against it too. and there's nothing stopping apple from sticking FLAC in an mp4 container with fairplay, we can't prevent that anyway. aside from the principle of it, another reasone we're against it in FLAC is that DRM is doesn't belong in the codec layer, it's a layer on top.
apple's got nothing to fear from FLAC, it can actually be used to their advantage to get a leg up on the competition, since for lossless electronic distribution FLAC is becoming the de facto standard.
FLAC - Free Lossless Audio Codec
Human being the key thing here. What makes you think that parent is human?
On the internet nobody knows you're a dog...
Lossless audio is going to involve some large file sizes, and with that, comes increased costs--bandwidth ain't free, and storage/delivery of these files is not going to be cheap or easy. This all translates into fairly expensive downloads.
So for Apple to seriously consider this, they're going to have to figure out if there are enough audiophiles out there willing to pay that kind of money for downloads.
Personally, I kinda doubt it.
Gifts for Geeks - Stuff that really matters!
i've gotten it the other way around... I've been told that i have 'the golden ear' when i told someone that they left there CRT tv on. i can hear the picture tube wine when it is displaying all black and no sound. maybe i do have an audiophiles ears, but i am certainly not a snob about it (those apple ear buds sure suck!)
i have trained eyes too. i see all sorts of compression artifacts on digital TVs that nobody else notices. i wish i could turn that ability off! or just bring back the analauge signal!
Don't call me back. Give me a call back. Bye. So yeah. But bye our, well, but alright we are on a shirt this chill.
Nyquist's theorem states that a wave of frequency f must be sampled at the rate of at least 2f in order for information not to be lost. So, yes, a 44.1kHz sampling rate can accurately reproduce signals up to 22kHz without loss of information, and since that's all we can hear, we should be fine. Right?
Well, not entirely. You see, if the source material contains frequencies above 22.05kHz, they will end up "aliased" onto another part of the frequency spectrum. In short, the extra high-end becomes noise. Information is lost.
Here is the important part, in practical terms. In order to prevent aliasing, the source material must be low-passed to remove the unrepresentable high frequencies. Low-pass filters are not perfect; in order to toss out the frequencies we don't want, we end up attenuating some of the frequencies we do want. Thus it is not uncommon for high-frequency rolloff to begin in the mid-teens of kilohertz, even though we're aiming for 22kHz as the corner frequency.
This causes a real, human-audible difference in the finished product, and it is practically impossible to avoid.
Now, with a 96kHz sample rate, we aim to squash all frequencies above 48kHz, and our non-ideal low-pass filter starts to work in the 30kHz range. The imperfections in the low-pass filter are only apparent at frequencies humans can't hear. The finished audio ends up sounding like the source material, with no human-detectable loss in fidelity.
This is why 96kHz is a good idea.
Cretin - a powerful and flexible CD reencoder
Easy, a square wave(or any wave) can be represented (through fourier transforms) as a sum of sine waves of increasing frequency. If you have a 22khz square wave, what you really have is a 22khz sine wave, and a bunch of sine waves with frequencies greater than 22khz. Those higher harmonics cannot be accurately represented with a 44.1 khz sampling rate, but since you can't hear anything above 22khz anyway it doesn't matter.
Give me Classic Slashdot or give me death!
You're hearing the horizontal scan, which is usually around 15kHz -- quite high, within an octave of our upper limit.
Cretin - a powerful and flexible CD reencoder
As much as it hertz, their loss results in your gain.
The theory of relativity doesn't work right in Arkansas.
Once, when my band was recording to a digital medium (a RADAR 24-track hard disk recorder, for those keeping score), we captured some tracks at 16bit, and some at 24bit. All other parameters in the signal chain were held constant.
I did not expect to hear as big a difference as I did. 24b absolutely crushed 16b in the oh-so-unscientific terms of listening enjoyment. Everything, especially the cymbals, sounded clearer, less harsh and brittle, more defined. We had to throw away some good 16b takes because they sounded so much worse than the 24b recordings.
Don't be so quick to discount the difference that a little extra dynamic range can make. Sure, you might not notice when you're listening to your iPod in your 89 Chevy Cavalier with the burned out left rear speaker, but it's not as hard to tell as you might think.
Cretin - a powerful and flexible CD reencoder
Very well put. It's one of the things that makes the delta-sigma modulation at very high sample rates used in eg SACD interesting. Of course, it would help if the data stream were easier to work with, which is why I think 24/96 or even 24/192 is superior overall.
The problem gets even more obnoxious if you care about the flatness and phase response of your filter. The one time I've done data acquisition work that cared about such things at 20kHz, we ended up using a 250kHz sample rate in order to give the Bessel filter room to operate. (We could have gotten away with marginally lower, but not enough lower to avoid buy the 1MS/s ADC system. We had 4 channels, so we ran at 250kHz.)
That is a good argument for why a studio should sample at a rate that accommodates the roll-off in their analog low-pass filters. However, once that is done you can use a can use a digital lo-pass filter / downsampler which can easily be designed to have very sharp cut-off rates. There is no reason at all for a consumer format to be more than 48kHz.
Meh, since when does the post -90s (or post 80s for that matter) music industry care about something as silly as dynamic range? You need to hear our music without having to turn up your radio! 3dB of dynamic range should be enough for anyone.
Nyquist's theorem states that you can accurately represent frequencies up to 1/2 the sampling rate. That is 100% true. But in the real world, if you are sampling a digital recorder at 44Khz how do you ensure that NOTHING above 22Khz gets to the analog to digital converter? You need a strong analog filter but there are no filters that have an exactly square cut off Maybe let's say you have a 24db per octave filter. This mens you will have only attenuated the higher frequencies, not eliminated them. Same on playback. You need a theoretically perfect analog filter to playback. Such analog filers do not exist. The way they get around all this is to sample at 96 or 128Khz. If you do this then real-world analog filters can be used.
No, it comes out as a sinewave, just as it went in. A triangle wave at 22.05 kHz is composed of sine waves at 22.05 kHz and then many higher harmonic frequencies. Only the fundamental will be reproduced of course.
Go read about Nyquist's Theorem before spouting falsehoods.
There is an advantage to higher sampling rates, but it has nothing to do with the frequency content of the recorded material or Nyquist's theory . If you sample at 44.1 khz (CD standard) you get 44.1 khz noise in the output. That has to be filtered out somehow, without affecting the in-band audio signal. Rolling off many DB in a short frequency span (factor of ~2) takes quite a filter, which depending on how it's done, introduces phase shifts of the in-band signal. The sound quality from CD players it largely determined by how, and how well, the D-A conversion (which has a frequency response all it's own determined by the guts of the converter)and analog filtering are done.
Sampling at higher frequencies makes it easier to build a good output filter. That's a very secondary or tertiary level effect, so it doesn't really make much difference, but it theoretically could.
Note that this is assuming the standard PCM encoding. "Single Bit"/streaming encoding (like SACD runs at fantastically higher sample frequencies, but the frequencies aren't really comparable (and it's not a good way to go because you introduce other issues (like tons of quantization noise).
The only identified issue with the standard red-book CD format is the dynamic range, but there are so few sources that need more than 16 bits and certainly very few playback systems/environments that will let you take advantage of it, it's essentially a non-issue. HDCD (which is a 20-bit PCM format) addresses this but hasn't and probably won't become common.
Bottom line - the guys who came up with the audio CD sampling format pretty well knew what they were doing and there aren't any practical limitations in the recording format. Everything else in the system (from microphone to engineering to speaker) is the limiting factor.
Brett
That said... the sampling frequency shouldn't be mixed with the signal frequency in the way you mention; e.g. 44.1KHz, divide by 2 (yay Nyquist), ~22KHz is the maximum frequency you can sample. ergo: 96KHz allows you to sample 48KHz signals and nobody can hear 48KHz anyway so what's the point.
Ah, true, but...
A 400Hz sine wave is now -also- sampled at the 96KHz level. Suddenly, that sine wave is looking twice as smooth.
No, it's not. If a 400 Hz sine wave is sampled at a mere 800 Hz, it can be reproduced perfectly, as long as your equipment is designed properly. Again, go read about Nyquist's Theorem before spouting falsehoods.
If you're reproducing that sine wave in a "blocky" way, your reproduction equipment is faulty. There's nothing wrong with the recording technique, because, again by Nyquist's Theorem, no information has been lost.
Personally, I'm beginning to wonder if the real reason for dynamic range compression is so that customers aren't surprised by how crappy some manufactured idol bands and singers sound in person without heavy studio voice processing.
Watch out, you're about to start an argument with all the people who think that it's normal for good bands to make albums with only one good song.
I wish I could find the link, but there's reasonable evidence from blind listening tests that people, though they would not necessarily report any quality difference, were able to report things about the recording like "I can tell the cello is sitting in front of the viola" and other things that are very subtle and spatial. This of course depends on headphones and careful binaural recording, so on most end products it wouldn't make much of a difference.
In my line of work, most sound designers are recording all of their sound effects at 96K and 192K, a bit for the quality (guns and loud transient stuff sound totally bitchin), but also because if gives a great deal more latitude when you want to pitch down something-- you don't hear 30K overtones on an explosion, but it's nice to have them there when you pitch the explosion down 2 octaves, and your 30K overtones are at 7.5K and help keep the sound from sounding like it came over a phone.
I know a lot of you on this thread are arguing that 24 bit is worth it on an end product, but remember that the effective dynamic range at 24 bit is around 120 dB, which exceeds your threshold of pain by about 10 dB, so you're getting a ton of dynamic range that you're just going to use the volume knob on in the end to flatten out. Also, that implies you're listening in a silent room. Your average city apartment or townhouse has a noise floor around 40 db SPL at least, so you'd better have acoustical treatments or be on headphones that isolate that much (the more a set of cans isolate, though, the worse their spectral character tends to be, though.
Don't blame me, I voted for Baltar.
Except you are presuming that the human ear perceives a 20kHz sine wave, and a 20kHz sine wave plus a whole series of harmonics identically.
The problem with that notion is that the standard test for hearing perception is to play pure sine waves of varying frequencies and ask the listener if they can hear them. However over the millions of years of human evolution, it was not until the invention of the tuning fork in 1711 that any human ear had heared a pure sine wave. Up until that point it had evolved to distinguish multiple frequencies at once.
I am not aware of any scientific studies into whether the human ear is able to perceive the existence of harmonics in sound waves above what is considered the normal hearing limit. Surprising really because if it is, it would explain a lot when it comes to sound.
True, but you're probably not going to hear all the details on your iPod anyway. The real advantage of this, at least in my mind, is the ability to transcode to a format you want without fear. I don't claim to be able to hear the difference between lossless and 192 or 256 MP3, but the idea of taking a 128 AAC and converting it to a 192 MP3 to play on something that doesn't support AAC is problematic, and that is something that can be fairly easy to hear.
So with hard drive sizes getting large enough that a moderate sized collection of lossless albums (say 100 Gb or 200 to 300 albums) isn't that ridiculous, so it makes sense to archive music in a lossless format. Of course I'm not sure if the bandwidth capabilities are really there yet to do this properly, but I'm no expert.
You're probably thinking of silver, not platinum. Silver actually conducts better than copper, but it's both more expensive and it suffers from corrosion. Gold is not a better conductor than either "out of the box", but after a bit of time and oxidation of those silver and copper contacts, it becomes better than both.
Long story short: a scientist will tell you silver is the best conductor, followed by copper, followed by gold. An engineer making something that needs to work more than a few days in the lab will note that gold is actually the best conductor of the three when put into real use, at least for those exposed contacts. Your silver or copper contacts are going to sound crappier than my gold ones all too soon...
"Convictions are more dangerous enemies of truth than lies."
> Those higher harmonics cannot be accurately represented with a 44.1 khz sampling rate, but since you can't hear anything above 22khz anyway it doesn't matter.
"The ear can't pick fundamental sounds at more than around 20 khz" != "the ear does a fourier transform and discard all harmonics above 22khz." The signal processing that a ear does to localize and identify sounds is a little more sophisticated.
I didn't do a double blind test, but even a seemingly small difference between a DAT recording at 44.1 and at 48 khz seems to make a slight difference in the sweetness of high end.
---- MISSING MISCELLANEOUS DATA SEGMENT --- [sigdash] trolololol
But that's not the point. I encode my CDs to FLAC, I can re-encode to any lossy or lossless codec I like without any degradation in quality. So it's perfect for archiving music. Or, indeed, buying downloads that I'm going to want to keep indefinitely. I see MP3s and other lossless codecs as something transient, an equivalent of cassette tapes - all right to listen to, but you wouldn't want to keep them forever.
You do know that most studios record on 16bit 48khz equipment, right? That 4khz doesn't make much of a difference. In fact, most studio masters are slap-dash affairs. Bad mikes, bad recording equipment, inadequate space, etc. All that crap puts all but the very best masters far below what CD Audio is capable of. In this real-world context, there is no point at all to formats like SACD and DVD-Audio. What people actually WANT is pretty clear. People want CDs with a 5.1 Dolby Digital mix, or an equivalent surround system. The studios have been highly-resistant to introducing such a format because it costs 10X as much money to master surround sound recordings as it does plain stereo.
I really don't understand all this audiophile crap. Most of the sources are so lousy there is little point in trying to optimize your equipment. This is sharply contrasted with videophiles, because the movie studios actually bother to master their DVDs (and before that, LaserDiscs) properly. The same is generally true of Blu-Ray and HD-DVD. pull it down to something that doesn't require DVD-type storage for a single album. This makes no sense. Lossy compression can introduce nasty effects and can kill your range. Even the best psychoacoustic models (like LAME) still have serious problems with certain tracks. An uncompressed album fits in 650 MB, far less than a DVD (9 GB). Using FLAC or a similar codec would get that down to about 350 MB, less than many of the video downloads on iTunes.
It's still a lossy format which strips out some of the audio detail. I'm no gold-connector-magnetically-balanced-shielded-cable audiophile, but I do appreciate being able to listen to the entire depth of a piece of music (especially classical).
Perhaps a better way of putting it would be 'the human ear cannot distinguish between 320kbps MP3 and FLAC if listened to on iPod headphones', which is fair enough. There's no need to include everything if all I'm going to do is listen to it on the bus. Which leads to my original point - MP3 is lossy. AAC which is my format of choice is better quality for the space and bitrate, but is still lossy. FLAC isn't, which means I could have my lossless FLAC copy on my desktop where there's easy storage space, then have iTunes automatically create reduced quality versions for carrying around on my iPod. Compression from lossless source is always better than compression from an already compressed copy.
Not to mention that the iTunes store *isn't* 320kbps. 128kbps for the normal content, 256kbps for iTunes Plus.
How many people can read hex if only you and dead people can read hex?
The RAW equivalent for audio would nice, but lossless would be what it would take for me (and everyone I know) to buy online.
If any of you remember cassettes, low end MP3s are about equal (IMHO).
I haven't bought / downloaded any music because of this factor - it's just not good enough when I can purchase the CD and deal with it from there.
AAC is pretty damn good, but no, I can tell the difference for the most part and well, really, come on, get real - they already SELL it lossless, it's not like you're twisting knobs to transfer it to the hard drive.
If anyone can get the majority of the Corporate Music above the line brain dead to listen, it'll be Jobs, and Team Apple, both of them.
~hylas
I definitely disagree about the dynamic rate in certain cases. (I'm an audio engineer, DSP programmer, and electro-acoustic musician) The problem with dynamic range is that you have the same number of bits to represent 0 to -6 Db (mono) that you do for -6 to -inf Db. Once you get down to the softest sounds, you often don't have many bits left at all to represent the sound; you only get the full range for the loudest of sounds, and the distribution of bits is linear, while the distribution of loudness is logarithmic. For most music, it's not a problem, but it does cause problems for orchestral works which can have a huge range of dynamics. For instance, in Messaien's Éclairs sur l'au-delà there's a bit with two triangle players on opposite sides of the stage playing rolls as soft as possible, and that never sounds right in recording. Also, there's plenty of electroacoustic pieces that benefit from the increased dynamic range.
If you record at 16 bit but allow 12 Db of headroom just in case of really nasty spikes (which can definitely happen with an orchestra), then you are now effectively recording with 16384 possible values instead of 65536, meaning only 84 Db of resolution with which to record. For most listeners, though 16bit/44.1khz is fine, and it absolutely destroys vinyl in terms of fidelity, aliasing, etc. be damned.
I know two audiophiles - professional audio mixers, to be exact, who absolutely have golden ears. They listened to a CD of an album master and frowned like something was wrong. The "image" wasn't right; "smeared" somehow. Turned out they could hear the difference between a master CD and a copy of the CD. The difference? Clock jitter. Yes, they could hear the effects of clock jitter. Both of these guys are legally blind which apparently sharpens other senses.
Most of the stuff on
This applies only to me, of course, as I was the only test subject however here are my findings:
I can stastically distinguish up to 256kbit/s MP3 vs FLAC.
At 320kbps my rate is around 65%, which is not sufficently higher than 50% to declare that it was distinguished (over my 20 tests).
The following equipment was used:
Sennheiser HD650
Benchmark DAC
Fed using Emu 1616 from computer.
Using my ZD5's from ZaphAudio (www.zaphaudio.com) which I built, I had less accuracy due to noise level in room.
Tests were done double blind using Foobar's ABX test application. Test tracks were Mel Torme - Sleigh Ride (Jazzy Christmas with Telarc), Herbert von Karajan - Beethoven's 9th (Mvmt 4) and Rachael Yamagata - Worn Me Down.
At least for me, FLAC is not by any means an absolute neccisary. The portability options for conversion to other formats is a huge factor looking forward however. I am sure that those with Ipod earbuds would have less resolution capacity, however my ears are not by any means extraordinary.