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Downgrading of kbps when converting WAV. to ALAC in iTunes?

post #1 of 5
Thread Starter 

I've read a great deal of threads on how you can convert WAV. to ALAC in iTunes without any difference in sound quality.

Some people stated that there is no loss of information by converting one Lossless file into another.

 

However, when I went and did this with some of my albums on iTunes (previously ripped in WAV. format from a CD) to save on space, the files went from all 1411 kbps to a range of around 800-1100 kbps.

 

There is no change in bit rate (16), and the size of the files have lowered as expected.

Just wondering if this is something to worry about, or if there is a solution to this problem.

 

This is my first post on Head-Fi, any feedback would be appreciated greatly!

Thanks! ~Tomahawke :) 

post #2 of 5

Welcome to Head-Fi! Simply put, there is nothing to worry about. 

 

If there were no bitrate loss for converting between pure WAV and a lossless compression algorithm, you would not get smaller files.

 

Bitrate can be thought of as another term for how much storage your music is taking up every second! It is not an inherent property of your audio recording unlike bit depth and frequency.

 

WAV is pure PCM data. Ever wonder how you arrived at 1411 kilobit/s? Do the math: 44100 KHz * 2 channels * 16 bits = exactly 1411 kbps uncompressed.

 

If a file takes up less space, it has to have a lower bitrate ;) But there is absolutely no quality loss from using a lossless algorithm. Try converting from ALAC back to WAV - it'll result in identical sound data and your old bitrate of 1411 kbps. It's the same as if you put your WAV file in a ZIP file and extracted it again - obviously the data gets smaller, but obviously the original signal can be recovered perfectly.

 

Why don't we use ZIP for compressing music? Because it simply doesn't work as well as ALAC and FLAC for sound, which were mathematically designed for compressing audio. 

 

Why does the bitrate of ALAC keep changing in the same track? Information theory says that more complex parts of the track (less entropy) can be compressed more easily than others. The lossless algorithm is more efficient at compressing certain parts of the music more than others (a sine wave, for example, contains barely any entropy at all, while a track containing purely random noise can't be compressed very much; in fact, the mathematics dictates that it can't be compressed at all!). Because the point of a compression algorithm is to save space, the encoder tries its best to minimize bitrate when it can - generally speaking, the 800 kbps parts of your music are mathematically less complex than the 1100 kbps parts! But since ALAC is lossless, by definition it is always preserving all the data. 


Edited by tninety - 12/21/13 at 11:06pm
post #3 of 5

It's just as simple as compressing a file. The bitrate goes down and so does the file size.

post #4 of 5

with no loss of usable bits. Basically knocking off the unused 0's withing the packets. 

post #5 of 5
Thread Starter 

Thanks for your quick responses! I was just a bit paranoid that I would lose SQ in exchange for the smaller size of the file, which isn't a problem as I have an 80GB iPod :D Thanks again!

 :beerchug: 

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