ipod download...

[anthonyTD]

hi all,
ipod question, ie, what's the best way to download your own albums onto it, what software to use to convert your own CD,s and music already on your computer to ipod.
come on you digi techs...
anthony...

[Filterlab]

iTunes, there's no other way to be honest. It's free for all platforms, simply download and install it and then import your CDs. Once imported, plug your iPod into the computer, wait for it to show in iTunes and then select the tracks you want on your iPod and slide them across.

As for converting the music, you don't have to. iTunes gives you the option of what file format you'd like to import in and the files stay in that format for the iPod. The iPod will happily play lossless formats too.

Oh, if you already have music on your computer, simply slide the files into iTunes and then slide them from there into your iPod.

[shane]

How do you deal with FLAC files? Do they have to be converted to AAC?

[Filterlab]

I have a Mac so I've never used FLAC, I used AIFF in the past and now I use CAF. There are programs to switch FLAC to other formats if required, but if Anthony is importing stuff fresh then it shouldn't be a problem. As far as I know the iPod isn't capable of handling FLAC files so they would have to be switched.

[Peter Stockwell]

Like Rob, I use a mac so don't use Flac.

If importing CDs from scratch set the action upon inserting a CD into the drive to "import and eject". Set the importing options to the compression rate that you desire, if you think it's just for the ipod 160 kbps AAC is way better than 128 kbps, and 192 and 256 are better than 160, but not by the same qualitative step, imo.

There's also an option that I find handy, it's "import missing artwork", I think it's in the advanced pane of the preferences, but without iTunes open I can't remember where it is exactly; The downside is that you have to have an iTunes account.

And another thing that's nice to know is, if you wnat your John Coltrane albums to show up in alphabetic order then in the Sorting tab of the get info pane enter "Coltrane, John".

and off you go.

[Beechwoods]

I use a Mac and have lots of stuff in FLAC format. Leopard plays FLAC natively now. Try playing it in Quicktime or Preview and it works. I've not tried it in iTunes. I use Cog as my main 'player' for FLAC playback on my Mac

There has just been a new major release of the Rockbox firmware which runs on iPods backs to 1st Gen, plus a host of other hardware (I use it on a Cowon iAudio M5L).

Rockbox is superb, there are loads of skins for it, and it plays all the major formats, compressed and uncompressed including FLAC, Shorten, MP3, Ogg, WAV, AIFF, AAC. Loading my machine with tunes is as simpler as copying files over to my USB connected player.

Check Rockbox out. It's fantastic.

[Filterlab]

I didn't realise that Cog played FLAC, although I have to say that CAF is an amazing file format, leagues ahead of anything I've heard. The architecture is rather clever to say the least and it shows Apple's willingness to keep them high in the professional and audiophile zones, something that's exceptionally lucrative. I know the Core Audio Architecture on Leopard is getting very trick now, more updates to come soon and next year OS 10.6 (Snow Leopard) is released which will switch the whole operating system away from depending on processor speed and over to multi-core operations, it'll remain in 64bit but with a potential for 16tb RAM!

Hopefully the Core Audio will allow for 64bit PCM rather than 64bit float, saying that there's little wrong with 64bit float (other than relying slightly on anti-aliasing), and 24bit and 32bit Linear are exceptionally precise word lengths as it is.

[Peter Stockwell]

Hopefully the Core Audio will allow for 64bit PCM rather than 64bit float, saying that there's little wrong with 64bit float (other than relying slightly on anti-aliasing), and 24bit and 32bit Linear are exceptionally precise word lengths as it is.

Is this engineer speak to say that the output digital stream will automatically adapt to the native sample rate and size of the file being played ?

[shane]

I have loads of MP3 and FLAC files which I normally play through a Squeezebox, but I just got an old Ipod to use in the car. I don't really want to convert a hundred gigs or so of FLACs, so I guess I have to restrict the Ipod to the MP3s?

[Filterlab]

Is this engineer speak to say that the output digital stream will automatically adapt to the native sample rate and size of the file being played ?

In a way. Basically a floating point operation has a greater mathematical range than a linear operation (a floating point operation can move its radix point to suit a data stream), however floating-point numbers achieve their increased range at the expense of less precision. Whereas a linear operation is bound to its sequence of amplitude values and quantized accordingly (word length or bitrate), the higher the quantization level the more accurate the reproduction of the original data.

So, whilst an audio file encoded in 64bit float may have a greater ability to vary an 'on the fly' data rate, it takes more processing power to decode and therefore compromises accuracy. The same track encoded in a linear word length (say 32bit) will be more accurate and easier to decode although it's word length is distinctly shorter. However at these word lengths it's almost negligible as 16bit is more than enough for audio reproduction and 16bit Linear PCM is the redbook standard, using greater word lengths simply improves accuracy as the more bits (times quantized) are used the greater the overall quantity of data that can be transmitted. In essence 64bit for two channel audio is being greedy (and it's simply a direct non-aliasing multiplication of 16bit), as a comparator Blu-ray will happily transmit eight channels of lossless audio within 24bit word lengths.

I'll give you an example;

A musician is playing a hi-hat cymbal at 16 times per bar (a 16th), if a sequencer then quantizes that instrument to 8ths (equivalent of 8bit) then every other cymbal sound disappears as it is quantized out. If the quantization is done at 16ths (equivalent of 16bit) then every cymbal sound is heard. However if that quantization rate is doubled to 32nds (equivalent of 32bit) then if any of the cymbal strikes are out of time slightly then the accuracy of the quantization picks the mis-timed strike up and moves it to the nearest 32nd of a measure. Doubling it again would then move it to the nearest 64th of a measure and so on. The higher the bitrate (times quantized), the more accurate to the original data the file becomes.