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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.
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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
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
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.
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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
There is also a Mac OS X FUSE version
http://code.google.com/p/macfuse/
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.
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.
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.
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.