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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."
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!
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
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
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
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.