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Re: Itunes, bitrate, and other questions...
pmbuko #284234 12/29/09 02:40 PM
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I find it interesting that there is no way to just highlight an album and export/copy it at a lower bitrate to your mp3 device, without leaving an extra copy on your hard drive. I will have to play around with it some more, thanks for the help..


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Re: Itunes, bitrate, and other questions...
SirQuack #284239 12/29/09 03:06 PM
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Eric, I just read an article about having more than one Library in Itunes. In windows, if you hold down the Shift key when launching Itunes, it gives you the option to create a different library, and which library you want to launch.

If I create a seperate library for my lower bitrate stuff, is there an easy way to create the copy into that new library from the other library?


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Re: Itunes, bitrate, and other questions...
SirQuack #284241 12/29/09 03:16 PM
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There isn't an easy way to copy files from one library to another. You can duplicate your entire music library, but then you still have to convert all the songs in one of them and throw out the lossless files.

And then, you'd have to manually manage new songs you import between the two libraries.

iTunes is nice, but it wasn't designed to do what you want it to.

Re: Itunes, bitrate, and other questions...
SirQuack #284243 12/29/09 03:37 PM
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I would not trust that option. With iTunes, the more you "see" what you do, the better. The playlist option is, as far as I know, the safest way to go.


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Re: Itunes, bitrate, and other questions...
EFalardeau #284244 12/29/09 03:43 PM
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I believe Media Monkey can Compress music on the fly when transferring it to the ipod.


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Re: Itunes, bitrate, and other questions...
Listener #284247 12/29/09 03:54 PM
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Thanks guys, sounds like keeping everything in one library and using "smart" playlists might be the best option. I guess I will have to have duplicates of every song at two different bitrates then...


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Re: Itunes, bitrate, and other questions...
ClubNeon #284257 12/29/09 04:47 PM
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 Originally Posted By: ClubNeon
 Originally Posted By: chesseroo
Just a general question.
I've had some discussions with some rather brilliant computer engineers and programmers (you know, the kind of guys who learned how to build circuit boards at the age of 12) simply stating 'any compression format can be assumed to induce errors by its very nature of how they compress data'.

I don't know what those guys were doing, but at 16 I was designing lossless packing algorithms for 2D graphics. I know for a fact that RLE (run length encoding) and dictionary pattern matching are lossless. They are also blindingly obvious, and a 16 year old could come up with them on his own.

Lossless audio is somewhat more complex, but I still have no doubt that it can be done without introducing error. One of the usual methods is to write an prediction algorithm which tries to guess what value will come next based upon previous. If it's right, then nothing needs to be stored. If it's wrong, then the difference between the predicted value and the actual value is stored. The error is the data.

Properly implemented, there are many ways to reduce the redundancy in data, that when undone presents an exact copy of the original. One thing, in packed form damage to the data will be more extensive when unpacked. But that's not a failure of the compression, but of the transmission/storage medium. And why we have error checking and correcting.

Unfortunately i cannot reproduce the conversation as i was merely a spectator and the language was beyond my knowledge base, but it does sound similar to what you are speaking of; the concept of prediction algorithms and how effective that is with the complexity of the audio range in a digital recording.

Does a 450.1 Hz tone get assigned the same compressed digital tag as a 450.001 Hz tone because they are 'close enough'?
Does a 560 Hz tone 1 second long get the same compressed hexadecimal digital tag as the 561 Hz note only 0.001 seconds long that follows it because the person writing the compression algorithm would 'know' that any note less than x seconds could not be interpreted by the human brain anyway?

This is a terrible example i realize, but if i understand their conversation correctly, that was the gist of the argument, that data could still be lost because of the predictive nature of the compression algorithms.


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Re: Itunes, bitrate, and other questions...
chesseroo #284265 12/29/09 05:11 PM
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I think it would really depend on how big the iPod was and how long you needed it to play for (or what variety you'd want). If you got a 8 or 16 GB iPod (not to mention one of the HD based models at 80 GB or so, who cares if you're using uncompressed music on there and the files are huge?

I keep all my music at at least 160Kbps, and now I do it at 256Kbps VBR AAC--and I use a 20 GB, a 30 GB, and a 2GB iPods. Plenty of music, even on the 2 GB one.


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Re: Itunes, bitrate, and other questions...
chesseroo #284276 12/29/09 05:27 PM
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With lossy encoding, what you're saying can sort of happen. MPEG audio uses a cosine transform. It basically takes the time-domain sample rate, and converts it into frequency domain. It than can analyze that frequency information, and try to find slices of a cosine wave which fit the original wave form. This Fourier transform in theory can be lossless, but it would take an infinite number of slices to completely reproduce an original analog signal.

Luckily CD audio has only 44,100 samples a second (time domain), and when converted into the frequency domain the highest frequency which can be reproduced is 22.05 kHz (Nyquist limit). So using 22k cosine slices for every second of audio would be able losslessly recreate the original PCM (within some very small margin of error). In practice that's a worse case, and one can get by with many less pieces and still have near lossless encoding.

But the problem is storing that many cosine coefficients would take more room than the original 2-byte (16-bit) samples. So in order to get the 10:1 compression which is common from MP3s, even fewer slices are used. This only gives a rough approximation of the original waveform. Fine details (like high frequency information) gets smeared together, and large rapid changes cause pre-echos or ringing.

None of this applies to truly lossless encoding schemes. They may use some perceptual estimation routine (DTS does this, Dolby's MLP along with FLAC use strictly numeric prediction) for the initial data reduction. But it goes one step further, and then checks this estimation, and looks back at the original recorded value. If there is any difference that's stored too. So as the final decode is done by the player/receiver it comes up with the estimated value, and then applies any correction to get the exact original data back. There's no loss or error.


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Re: Itunes, bitrate, and other questions...
ClubNeon #284282 12/29/09 05:31 PM
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Yeah, if it's already in the digital domain, it's easy to compare the compressed with the uncompressed on an exact level--the computer can just uncompress it to check!

As long as it's in the digital domain, the computer doesn't think about it in terms of length of the note or tone--it's all just bits.


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