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