Hi all.
I have some high bitrate lame encoded mp3's, and would like to convert them to a lower bitrate for a flash player. However, I'm confused as to why lame adds/deletes samples upon recoding.
For example, a 320kbs cbr file was rencoded using the same settings. After decoding both files to wave - again with lame - the EAC compare tool showed the first file was missing 1105 samples from the beginning compared to the second file. If I've read the results correctly that means the second file had been padded at the front. Why does this happen? I thought lame wrote info about encoding delays etc into the files. Is it just not using them?
This is no big deal for me. But I'm curious as to how and why this happens.
