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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.
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!
You might like all the songs. Since I listen mostly to rock/alternative genres, I probably only want his singles. Can I buy them cheaply, like my dad could in the 70's? He's got a stack of 45's that probably reaches the ceiling of my apartment.
I do realize they still sell them, but are they $0.99 per song cheap?
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
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.
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...
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
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
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.
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.
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
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